Home

Avaya IP Telephony User's Manual

1. Cisco 1600 Cisco 3600 Win 98 with NT with Chariot Endpoint 2 Chariot Endpoint 1 and Sniffer and Sniffer Parascope WAN probe NetIQ Chariot v4 0 was used to simulate VoIP calls between the two endpoints Chariot v4 0 accurately simulates the characteristics of various codecs and uses a 40 byte IP UDP RTP header Sniffer Pro v3 50 02 was used to capture the sent and received packets The Cisco 3600 had IOS v12 1 2 T and the Cisco 1600 had IOS v12 0 12 The Fredericks Engineering Parascope WAN probe was tapped into the V 35 serial link to take bandwidth measurements This test was performed using PPP encapsulation on the WAN link A single call was placed between the Chariot endpoints using the two most common codecs sending 20 ms voice packets Below are the results with and without RTP header compression Note that these are rough measurements Codec Payload Packets sec Avg WAN BW consumption kbps reduction bytes packet G 711 64 kbps G 729A 8 kbps For each codec there was an attempt to verify that the audio packets were received in tact This was done by spot checking the audio packets before and after compression using two Sniffer protocol analyzers With G 729 the RTP header and payload were identical before and after compression With G 711 however the received packets had the PADDING flag set in the RTP header although the flag was not s
2. Gives status of signaling group Gives status of trunk group or group member Gives Ethernet interface in out statistics for an IP board Gives C LAN board statistics multiple pages Gives C LAN socket usage Gives MedPro MR320 status for all boards or individual board Gives status of inter region connectivity Lists all administered speed duplex settings for IP boards Gives administered vs actual speed duplex settings for a board Sends pings and trace route from a board or from a station If board specify board lt slot gt If station specify source lt port gt where port is from status station form Use Help feature Table 7 Common SAT troubleshooting commands KW Avaya IP Telephony Implementation Guide 45 4 Guidelines for Avaya 4600 Series IP Telephones This section covers some general information regarding various Avaya 4600 Series IP Telephone models More specific information is available in the 4600 Series IP Telephone LAN Admiunistrator s Guide and other IP telephone guides at support avaya com The current GA firmware releases can be obtained at the same site Be sure to read the readme files that accompany each firmware package Note For simplicity in many IP telephone applications a C LAN is often called a gatekeeper although the call server is the gatekeeper and the C LAN is only a front end to the gatekeeper 4 1 Basics Legacy Models vs Current Models Legacy Avaya IP Tele
3. L IP UDP RTP Voice Payload E2 header 20B 8 B 12 B Variable trailer Figure 12 L2 overhead Ethernet Overhead G 711 20 ms call over Ethernet 90 4kbps Ethernet has a header of 14 bytes and a trailer of 4 G 711 30 ms call over Ethernet 81 6kbps bytes It also has a 7 byte preamble and a 1 byte G 726 20 ms call over Ethernet 58 4kbps start of frame delimiter SFD which some G 726 30 ms call over Ethernet 49 1 kbps bandwidth calculation tools do not take into G 729 20 ms call over Ethernet 34 4kbps consideration Nevertheless the preamble and SFD G 729 30 ms call over Ethernet 25 6kbps consume bandwidth on the LAN so the total Ethernet overhead is 26 bytes When transmitting 20 ms voice packets the Ethernet overhead equates to 10 4kbps 26 8 50 which must be added to the 80kbps for G 711 40kbps for G 726 and 24kbps for G 729 For full duplex operation the totals are 90 4kbps for G 711 50 4kbps for G 726 and 34 4kbps for G 729 For half duplex operation these figures are at least doubled but effectively the load is higher due to the added overhead of collisions WAN Overhead The WAN overhead is calculated just like the Ethernet overhead by determining the size of the L2 encapsulation and figuring it into the calculation L2 headers and trailers vary in size with the protocol being used but they are typically much smaller than the Ethernet header and trailer For example the PPP overhead is only 7 bytes However
4. DHCP requests ip helper address lt P addr of DHCP server gt to the DHCP server if the server is on a different subnet b Follow the instructions at the end of appendix A to get the IP phone on the auxiliaryvlan voice VLAN c After the phone boots up press Hold ADDR to verify that the phone received an IP address and associated information for the auxiliaryvlan 3 For call servers IP boards ie C LAN and MedPro MR320 and other VoIP resources configure their ports on the Eth switch to be native to the auxiliaryvlan That is these ports do not require both KW Avaya IP Telephony Implementation Guide 63 a port native VLAN and an auxiliaryvlan Just make the auxiliaryvlan the port native VLAN on these ports set vlan 200 lt mod port gt assuming 200 is the auxiliaryvlan ID Then disable the auxiliaryvlan feature on these ports set port auxiliaryvlan lt mod port gt none 4 Always verify network connectivity between devices using pings and trace routes KW Avaya IP Telephony Implementation Guide 64 Appendix C RTP Header Compression RTP header compression is a mechanism that reduces the protocol overhead associated with VoIP audio packets It is a function of the network and not a function of the VoIP application Along with the benefits of using RTP header compression there are also cautions and this appendix discusses both Application Perspective Here is the anatomy of a 20 ms G 729 audio packet which 1s rec
5. payload must be divided into three fragments of 80 bytes 80 bytes and 20 bytes Each fragment incurs a 20 byte IP header to make the packets 100 bytes 100 bytes and 40 bytes A single 200 byte IP packet must be fragmented into three separate IP packets to meet the 100 byte MTU In addition the L2 overhead also increases because each L3 packet requires L2 encapsulation MTU should not be an issue for VoIP because most interfaces have a default MTU of 1500 bytes However it is possible to intentionally set the MTU to low levels Even if the MTU is not set to a level that would fragment VoIP packets the principle of fragmenting the L3 payload and incurring additional L3 and L2 overhead applies universally Changing the MTU requires a thorough understanding of the traffic traversing the network A low MTU value relative to any given IP packet size always increases L3 and L2 overhead as illustrated with the VoIP example Because of this inefficiency it is generally not advisable to lower the MTU L2 Fragmentation The second factor involves fragmenting the L2 payload or the entire IP packet This process of fragmenting a single IP packet into multiple L2 frames incurs additional L2 overhead but no additional IP overhead For example the fixed cell size for ATM is 53 octets bytes with 5 octets for ATM overhead and 48 octets for payload Without header compression there is no way to get a voice packet to fit inside one ATM cell Therefore the L3
6. to allow for a high margin of error assume a 14 byte total L2 encapsulation size which would add an overhead of 5 6kbps 14 8 50 assuming 20 ms voice packets This would result in approximately 43kbps G 729 20 ms call over PPP 26 8kbps for G 726 and 30kbps for G 729 over a WAN link G 726 20 ms call over PPP 50 8kbps Significant bandwidth savings are realized by using a compressed codec G 729 recommended across a G 729 20 ms call over 14 B L2 29 6kbps WAN link Note that in today s data networks most if G 726 20 ms call over 14 B L2 53 6kbps not all WAN links are full duplex L3 Fragmentation MTU Related to bandwidth there are two other factors that must be considered for operation across WAN links and they both involve fragmentation The first factor maximum transmission unit MTU involves fragmenting the layer 3 L3 payload The MTU is the total size of the L3 packet IP header IP payload which is 200 bytes for G 711 and 60 bytes for G 729 assuming 20 ms voice packets If the MTU on an interface is set below these values the IP payload UDP RTP voice payload must be fragmented into multiple IP packets each packet incurring the 20 byte IP overhead For example KW Avaya IP Telephony Implementation Guide 19 suppose the MTU on an interface is set to 100 bytes which is an extremely low value The 20 ms G 711 IP packet is 200 bytes consisting of a 20 byte IP header and a 180 byte IP payload The 180 byte
7. CLI The MGP requires a VLAN IP address and mask These are displayed and configured using the MGP CLI commands show interface mgp and set interface mgp type configure to enter configuration mode for the set commands The MGP may be on the same VLAN as the inband interface or on a different VLAN If ona different VLAN a L3 router is required to route between the two VLANs Like the inband interface the MGP also needs at least a default route to route off of its VLAN subnet The MGP CLI commands are show ip route mgp and set ip route to display and configure MGP routes Each VoIP media module also requires an IP address using the set interface voip v command The VoIP modules inherit the VLAN mask and configured routes of the MGP so there is no need to explicitly configure them for each VoIP module The internal VoIP module is voip v0 An external VoIP module would be voip v1 or voip v2 or voip v3 or voip v4 depending on which slot it is in show mm shows all the media modules and their slot numbers G700 802 1p Q and DSCP The G700 can receive its audio and call signaling priority values from the call server s ip network region form or from local configuration The MGP CLI command show qos rtcp shows the locally set values and the values downloaded from the call server along with which set of values is in effect The command set qos control determines which set of values is used The simplest implementation is to use the values from th
8. Ethernet0O EthernetO Router SerialO WAN SerialO Router 192 168 1 0 24 192 168 2 0 24 other subnet other subnet Example 1 Ideal WAN terminating on G700 G350 G250 gateway Suppose all endpoints are capable of marking with one DSCP for audio and another DSCP for signaling This would be true in an Avaya Communication Manager system with TN799DP C LAN boards running firmware v5 or later Previous firmware versions and the TN799C board cannot mark at L2 or L3 A matching set of configurations is applied to both routers class map match any voipAudio create a class map called voipAudio match ip dscp 46 any packet with DSCP 46 is in this class class map match any voipSig create a class map called voipSig match ip dscp 34 any packet with DSCP 34 is in this class class map match any ipsiSig create a class map called ipsiSig match ip dscp 36 any packet with DSCP 36 af42 is also in this class policy map voipQoS create a policy map called voipQoS class ipsiSig give packets in the ipsiSig class 128K bandwidth 128 of this WAN link class voipAudio reserve 768k of this WAN link for packets in the vorpAudio class priority 768 class voipSig reserve 48k of this WAN link for packets in the voipSig class bandwidth 48 class class default put everything else in the default class and transmit it out the default fair queue queue in a weighted fair queue fashion random detect dscp based if the default queue starts to get full random
9. Guide Remote gateways and stations are controlled by the S87xx servers via the C LAN boards The remote 8300 is in local survivable processor LSP mode to take over as call server if 13 Trunks QSIG H 323 Q 931 e DCP ee errs F Manager A E y Call S8300 G700 fi asiGorpcs PCP H 323 Q 931 H 225 IP IP PSTN Public Switch G650 Public Switch QSIG or DCS ve Geet eT PRI PRI I P Public Switch Loop Start Inband HU T1 I I AS QSIG IIL IIL Q 9314 Analog VendorX PBX PRI SEFINITY System Figure 9 Trunks This figure illustrates a broad picture to put trunks into context PSTN trunks use the Signaling System 7 SS7 signaling protocol This protocol is not relevant to private enterprise telephony systems Private systems such as the IP Connect and DEFINITY servers in this illustration commonly connect to public switches using ISDN PRI trunks with Q 931 signaling Two private systems commonly connect to one another using T1 trunks with inband signaling or ISDN PRI trunks with Q 931 signaling This 1s illustrated in the trunks connecting the DEFINITY server to the P Connect and to the Vendor X PBX QSIG is a standard feature rich signaling protocol for private systems and it rides on top of Q 931 as illustrated between the DEFINITY server and Vendor X PBX DCS is the Avaya proprietar
10. IP UDP RTP Headers for Low Speed Serial Links www ietf org February 1999 KW Avaya IP Telephony Implementation Guide 81
11. MR320 The MR320 supports up to 320 calls based on licensing as opposed to a fixed max of 64 calls for the MedPro These boards perform the conversion between TDM and IP The audio payload is encapsulated in RTP then UDP then IP KW Avaya IP Telephony Implementation Guide 9 Multi Connect Adjunct Location Medium Large Enterprise Main Location H 225 RAS amp z Q 931 signaling ii iis L2 switch L2 ey Control t IP Network E i i CCMS over H 225 TCP IP PN PN PN RTP i nin nin aQ aQ i C LAN di IP Net C LAN iii MedPro MedPro RTP E PN C z Enterprise F No F pany A TDM bus a a hin i IP Network DCP MCC SCC ST MCC DCP k a amp CCMS and bearer Center Stage amp Raa nee k over TDM or ATM _ or ATM PNC pasg Figure 3 Multi Connect Multi Connect is the same underlying DEFINITY architecture except that the processor boards are replaced with much more powerful Avaya S8700 or S8710 Media Servers Port networks get IP Server Interface IPSI boards to communicate with the S87xx call servers CCMS exchanges between the call servers and port networks now take place over the control IP network Not all port networks require IPSI boards because Center Sta
12. SA is a client software application used to access the SAT interface on all Avaya servers Additionally on all but the DEFINITY servers SAT can also be accessed by telnet ing to the server 3 1 S87xx S8500 Servers The S87xx and S8500 are 19 inch rack mountable Red Hat Linux server platforms S87xx servers operate in a redundant pair whereas the S8500 is a simplex server Each server is configured and managed via a variety of web interfaces with the Maintenance Web Interface being the most comprehensive The web interfaces are designed to facilitate all anticipated configuration and management requirements and there is little or no need for a customer to access the Linux shell In an S87xx pair one of the servers is active and the other is standby SAT administration is performed on the active server and it 1s automatically carried over to the standby server Either of the servers could be active or standby at any given time and there are different ways to determine which is active If the two servers are on the same subnet there is a virtual IP address which is labeled the active server address in the Configure Server Configure Interfaces screen of the Maintenance Web Interface Whichever server is active takes control of the active server address and telnet ing or browsing or pointing Avaya SA to that address accesses the active server If the two S8700 servers are not on the same subnet server separation there is no virtual active serv
13. TN2602AP circuit packs and they can be administered for duplication Duplicated TN2602AP circuit packs will operate in an Active Standby mode State of health parameters exist between the two boards to determine when it is appropriate to interchange duplicated TN2602AP circuit packs The failover from Active to Standby can take up to 3 seconds depending on the type of fault without interruption of service Duplicated TN2602AP circuit packs in a PN share a virtual IP and virtual MAC address These virtual addresses are owned by the currently active TN2602 In addition to the virtual IP address each TN2602 has a real IP address All bearer packets sent to a PN that contains duplicated TN2602AP circuit packs regardless of whether the packets originate from TN2602s in other PNs or from IP phones or gateways are sent to the virtual IP address of the TN2602 pair in that PN Whichever TN2602AP circuit pack is active is the recipient of those packets When failover to the KW Avaya IP Telephony Implementation Guide 34 standby TN2602 occurs a negotiation between TN2602s to determine which TN2602 is active and which is standby takes place State of health call state and encryption information is shared between TN2602s during this negotiation The newly activeTN2602AP circuit pack sends a gratuitous address resolution protocol ARP request to ensure that the LAN infrastructure is updated appropriately with the location of the active TN2602 Other devices wit
14. TN799DP C LAN board is 10 half to make it backwards compatible with the previous TN799C board which could only do 10 half When a C LAN or MedPro MR320 is inserted into one of the port networks the board receives its speed duplex programming from Communication Manager per the appropriate form If for any reason a board loses this programming it reverts back to the board s default The maximum throughput for a MedPro board is DO e VLOGO Cioe pU e 5 8Mbps which is what is required for 64 G 711 20 5 8Mbps for 64 G 711 20 ms calls ms calls over Ethernet The maximum throughput for a C LAN board is much less than this Therefore the minimum speed duplex requirements are 100 half for 29Mbps for 320 G 711 20 ms calls the MedPro and 1O half for the C LAN Due to its high capacity the MR320 board should always run at 100 full If there is poor audio quality on calls going through a particular MedPro MR320 board follow these steps to determine if a speed duplex mismatch between the MedPro MR320 and the Ethernet switch is the cause Check both the board get ethernet options lt slot gt and the Ethernet switch port and verify that they are set to the same speed duplex or have auto negotiated to the same speed duplex Check for L1 errors as instructed in section 2 1 under the Speed Duplex heading Send a continuous ping ping t to the MedPro MR320 from a Windows machine If the pings intermittently fail and the failures coincide wit
15. These values can be configured locally via the set qos bearer signal commands or they can be downloaded from Communication Manager On Communication Manager these values are configured on the SAT ip network region form for the region to which the gateway is assigned A gateway is configured to use the Communication Manager values by executing set qos control remote on the MGP gateway CLI The CLI command show gqos rtcp displays the locally set and remotely downloaded values as well as which values are in use The G700 G350 and G250 gateways are all administered in Communication Manager via the SAT media gateway form which is covered in section 3 5 3 4 G650 G600 MCC1 and SCC1 Gateways Port Networks The G650 G600 MCC1 and SCC1 are non H 248 media gateways They are controlled via the Avaya CCMS protocol unlike the G700 G350 G250 gateways which are controlled via the H 248 protocol The CCMS based gateways are better known as port networks and they share the same port boards The most significant boards related to IP telephony are the C LAN TN799DP MedPro TN2302AP and MR320 TN2602AP boards Boards with these specific codes are required for Communication Manager previous board revisions cannot be used C LAN Capacity and Recommendations The Control LAN C LAN board is the IP interface for many functions including H 225 call signaling for IP stations and IP trunks H 248 media gateway control signaling connectivity to vari
16. all permits all VLANs and not just the configured 3 Ifthe port is connected to a router or to another switch trunking must be enabled with the command set trunk lt mod port gt dotlq which causes all egress frames to be tagged However if the port is connected to an Avaya IP phone with an attached PC trunking must not be enabled so that none of the egress frames are tagged This is necessary because most PCs do not understand tagged frames Setting the Priority without Trunking or VLAN binding Single VLAN Scenario With Avaya switches it is possible to set the L2 priority on the IP phone even if the phone is not connected to a trunk or multi VLAN port That is the Avaya switch does not need to be explicitly configured to accept priority tagged Ethernet frames on a port with only the port native VLAN configured This is useful if the phone and the attached PC are on the same VLAN same IP subnet but the phone traffic requires higher priority Simply enable 802 1Q tagging on the IP phone set the priorities as desired and set the VID to zero 0 Per the IEEE standard a VID of zero assigns the Ethernet frame to the port native VLAN Cisco switches behave differently in this scenario depending on the hardware platforms and OS versions Here are Avaya Labs test results with a sample of hardware platforms and OS versions Catalyst 6509 w Accepted VID zero for the native VLAN when 802 1Q trunking was CatOS 6 1 2 enabled on the por
17. be applied to the call server when communicating with this IPSI board values are not applied to the IPSI board itself The IPSI s speed duplex and L2 L3 priority values are configured on the board itself instead of via SAT forms From the IPSI board type ipsilogin at the IPSI prompt and enter the login name and password to access the IPADMIN prompt The commands to display and configure the control port speed and duplex are show port 1 set port negotiation 1 set port speed 1 and set port duplex 1 The commands to display and configure the L2 and L3 priority values are show qos set vlan tag set vlan priority and set diffserv Be sure to understand what these values do before setting them see all of section 2 3 particularly the heading Rules for 802 1p Q Tagging 3 5 General IP Telephony Related Configurations SAT Forms The SAT interface has various forms that are used to configure specific features This section covers the forms used to configure general IP telephony Most of the forms have a display option to view the current configurations and a change option to change them Some also have a list option to view for example a broad list of stations without seeing in detail how each station 1s configured ethernet options As of Avaya Communication Manager 2 0 each IP board s speed and duplex settings are configured using the ip interface form The ethernet options form has the list and get options to verify
18. changes the size and format of the comprehensive Ethernet and 802 3 frames Because of this many intelligent switches ones that examine the L2 header and perform some kind of check against the L2 frame must be explicitly configured to accept 802 1Q tagged frames Otherwise these switches may reject the tagged frames The Tag Protocol Identifier TPID field has hex value x8100 802 1QTagType This value alerts the switch or host that this is a tagged frame If the switch or host does not understand 802 1Q tagging the TPID field is mistaken for the Type or Length field which results in an erroneous condition DIX Ethernet v2 frame Dest Addr Src Addr Type Data pi 6 octets 6 octets 2 46 to 1500 octets 4 eee 802 3 Ethernet frame DSAP Prtcl ID or Frm Chk Dest Addr SrcAddr Len gong Org Code Data S n 6 octets 6 octets 2 Control Ethertype 38 to 1492 octets 4 octets 802 3 heade 802 2 802 2 LLC SNAP 8 octets total 802 1p Q TPID Priority i VLAN ID 2 octets 3 bits O 12 bits TCI 2 octets Figure 13 802 1Q tag The two other fields of importance are the Priority and Vlan ID VID fields The Priority field is the p in 802 1p Q and ranges in value from 0 to 7 802 1p Q is a common term used to indicate that the Priority field in the 802 1Q tag has significance Prior to real time applications 802 1Q was us
19. differentiated and segregated into various classes The term Quality of Service refers to what the network does to the marked traffic to give higher priority to specific classes If an endpoint marks its traffic with L2 802 1p priority 6 and L3 DSCP 46 for example the Ethernet switch must be configured to give priority to value 6 and the router must be configured to give priority to DSCP 46 The fact that certain traffic is marked with the intent to give it higher priority does not necessarily mean it receives higher priority CoS marking does no good without the supporting QoS mechanisms in the network devices CoS 802 1p Q at the Ethernet layer L2 and DSCP at the IP layer L3 are two CoS mechanisms that Avaya products employ These mechanisms are supported by the IP telephones and most IP port boards In addition the call server can flexibly assign the UDP port range for audio traffic transmitted from the MedPro MR320 board or VoIP media module Although TCP UDP source and destination ports are not KW Avaya IP Telephony Implementation Guide 20 CoS mechanisms they are inherently used to identify specific traffic and can be used much like CoS markings Other non CoS methods to identify specific traffic are to key in on source and destination IP addresses and specific protocols ie RTP 802 1p Q The figure below shows the IEEE 802 1Q tag and its insertion point in the Ethernet and 802 3 frames Note that in both cases the 802 1Q tag
20. explicit RAS Unregistration Request URQ message but it considers itself unregistered from that gatekeeper and is moving on to the next Even if the phone did send a URQ chances are the gatekeeper would not receive it because the failure condition could still exist The final retry interval prior to discovering would appear to give extra time for the failure to recover And indeed if the phone did receive a KA acknowledgment within that final retry interval it would stay registered to the same gatekeeper However the reality is that if the phone doesn t receive an acknowledgment within a second or two after the final retry KA it won t receive one Therefore the final retry interval really does not factor into the time to unregister Time to unregister answers the question How long must the failure ie network outage last before the IP telephone unregisters If the failure recovers just before the final retry KA is sent the phone remains registered to the same gatekeeper If the failure recovers a couple seconds after the final retry KA is sent the phone most likely unregisters and moves on to the next gatekeeper after the final retry interval KW Avaya IP Telephony Implementation Guide 50 The TCP and RAS keepalive algorithms are as follows IP telephone TCP KA TCP KA Time to RAS KA RAS KA Time to regular intrvl retry intrvl unregister regular intrvl retry intrvl unregister 4620 10 20sec 5 5sec 25 to 45sec ob
21. flow because the serialization delay is low and because there are so few signaling packets relative to audio packets Also packet loss should not be an issue because the queue should be large enough to sustain both audio and signaling Suppose however that the ratio of signaling to audio is much greater perhaps nearly 1 1 This would be possible in a remote office where all the signaling goes to a main office but most of the audio is local Suppose also that the WAN link is relatively small typically less than 768k and serialization delay is a factor In this case a large signaling packet entering the priority queue could delay audio packets and even induce packet loss if the WAN link and thus the priority queue are small enough It would be advisable in this case to use separate queues optimized for the different characteristics of audio and signaling As stated previously the X330WAN router and G350 G250 integrated routers are optimized to use separate queues for audio DSCP 46 and signaling DSCP 34 or 41 KW Avaya IP Telephony Implementation Guide 72 The preceding paragraphs are generalizations and are not meant to imply a firm set of rules Queuing is very complex and implementations vary among manufacturers The explanations given here are intended to give the reader a starting point Testing with live traffic and real equipment coupled with some trial and error will ultimately dictate the optimum configuration for a given p
22. form it is used for call routing purposes see the Avaya Communication Manager Network Region Configuration Guide at www avaya com Site Data can be used to note the gateway s address ie if it is located at a remote branch office For G250 models Max Survivable IP Ext refers to how many IP stations are permitted to fail over to the gateway when connectivity to the primary call server is lost This is part of the SLS feature new to Communication Manager 3 0 and the G250 The remaining information 1s automatically populated when the gateway registers with the call server system parameters mg recovery rule Options are change and display When a media gateway loses connectivity to the primary call server it can fail over to an LSP This form new to Communication Manager 3 0 administers rules that determine when a media gateway automatically recovers back to the primary server The Number is simply a numeric index Rule Name is a text descriptor Migrate H 248 MG to primary and Minimum time of network stability are the two conditions that must be met before the primary Communication Manager server accepts a media gateway recovery registration First the minimum network stability time condition must be met Then the recovery can happen Immediately When there are no active calls on the media gateway During a specified time window Either when there are no active calls or during a specified time window A blank Migrate
23. is sent to the Ethernet switch uplink port with the tag but to the attached PC user port without the tag This also allows the attached PC to communicate with the IP telephone when they are on the same VLAN and the phone is tagging KW Avaya IP Telephony Implementation Guide 46 DHCP Option 176 Just the basics of DHCP option 176 are covered here See the 4600 Series IP Telephone LAN Administrator s Guide for more details The DHCP specification has what are called options numbered from 0 through 255 Each option 1s associated with a specific bit of information to be sent by the DHCP server to the DHCP client For example option is the subnet mask option and is used to send the subnet mask to the client Option 3 is the router option and is used to send the default gateway address and other gateway addresses to the client Some options are defined such as options 1 and 3 and others are not The defined options are found in RFC 2132 Options 128 through 254 are site specific options They are standard options that are not defined and vendors may use these options and define them to be whatever is necessary for a specific application Avaya IP telephones use site specific option 176 as one of the methods to receive certain parameters from the DHCP server For the Avaya application of option 176 it is defined as a string The string contains parameters and values separated by commas as illustrated after the following table T
24. network region Options are change display and list This form is used to define the characteristics of an Avaya Communication Manager network region While this section describes the configuration parameters of the ip network region form the overall explanation of network regions and guidelines for network region design are covered in detail in the Avaya Communication Manager Network Region Configuration Guide at www avaya com The Location parameter is used to assign IP stations in this network region to a specific geographic location identifier KW Avaya IP Telephony Implementation Guide 38 The Authoritative Domain applies to Session Initiation Protocol SIP applications which are not covered in this document The Name is an arbitrary string to describe the network region The Codec Set refers to one of the seven codec sets defined using the ip codec set form and specifies which codec s are used by the endpoints in this network region The UDP Port Min Max is the range used for RTP audio by the MedPro and MR320 boards and VoIP media modules in this network region Use the following points to configure a more narrow UDP port range to set up security filters for example 2048 is the beginning of the range by default but this can be changed to a higher starting point It is recommended to use UDP ports outside the range of reserved ports A starting port of 50000 is outside the range of any reserved ports The MedPro
25. not be enabled without the supporting IP network configurations These configurations can be cumbersome and require a significant amount of network overhead A better call admission control CAC mechanism 1s native to Communication Manager as of 2 0 and is explained in detail in the Avaya Communication Manager Network Region Configuration Guide at www avaya com The H 323 Link Bounce Recovery parameters the LSP list on page 2 of this form and the inter region connectivity matrix beginning on page 3 of this form are covered in detail in a separate document See the Avaya Communication Manager Network Region Configuration Guide at www avaya com Inter Gateway Alternate Routing IGAR on page 2 of this form is a new feature for Communication Manager 3 0 This feature is covered in detail in the Avaya Communication Manager Network Region Configuration Guide at www avaya com Related to IGAR is a new parameter on the cabinet form to assign the cabinet to a network region The assignment of a cabinet to a network region which is a concept new to Communication Manager 3 0 applies primarily to IGAR It has no relation to IP boards in that cabinet and it does not assign traditional resources attached to that cabinet such as non IP stations and trunks to a network region ip network map Options are change and display This form is used to assign stations to Communication Manager network regions by IP address range or subnet Ifa stati
26. not to give false indications The results of this trace route are logged on the call server with an IPEVT tag one of many events with that tag H 248 Media Gateway and H 323 IP Endpoint See the Avaya Communication Manager Network Region Configuration Guide at www avaya com for information on most of the parameters under these headings Only the Periodic Registration Timer 1s covered here This timer determines the frequency at which a forcefully unregistered IP phone attempts to re register The primary application 1s for a desktop IP telephone that is forcefully unregistered because a user from home takes over the extension with a softphone At some point the user logs off the softphone leaving the extension free for the IP phone to reacquire However the IP telephone doesn t know when the softphone logs off so the IP phone simply attempts to register periodically and succeeds only after the softphone logs off This timer determines that frequency and it requires IP telephone 2 1 or later Music on Hold This feature applies to media gateways and to port networks in IP Connect systems with no traditional PNC Center Stage or ATM When music must be delivered via IP between media gateways and port networks the music should be transported via the G 711 codec for quality reasons If network region assignments are such that there is always a G 711 path between media gateways and port networks this feature is not necessary In some conf
27. of the G700 to the other devices is another factor A G700 s primary role is that of IP telephony specifically media conversion A P330 switch s primary role is that of L2 switching processing and forwarding Ethernet frames managing broadcast domains VLANs participating in Spanning Tree etc Depending on the implementation especially if there are no dependencies between the G700 and the P330 stack 1t may be prudent to keep the two roles separate so that a problem with either the G700 or the P330 stack does not adversely affect the other These points are mentioned to provoke thought in design and implementation Whatever the decision a G700 can fully participate in Octaplane stacks with other G700s or with P330 switches Bandwidth is another key factor for using or not using the Octaplane stack The G700 components P330 inband MGP VoIP modules S8300 require a certain amount of bandwidth to communicate off the chassis Each VoIP module consumes a maximum of approximately 6Mbps to service 64 G 711 calls using 20 ms packets With up to five VoIP modules on a single G700 the maximum bandwidth consumption is approximately 30Mbps Other than firmware and translation downloads the bandwidth KW Avaya IP Telephony Implementation Guide 30 consumed by the other components is negligible Therefore a single 100M uplink from EXT1 or EXT2 to another Ethernet switch is sufficient for the G700 itself The added bandwidth of the Octaplane s
28. on older Catalyst 6500 code pre 5 5 14 6 3 2 7 2 2 when the port is in trunk mode The resulting request was to use auxiliaryvlan instead of explicit trunking because portfast can be enabled on auxiliaryvlan ports even on the older code releases Interoperability with auxiliaryvlan and voice vlan was successfully lab tested on the following platforms with no known issues to date auxiliaryvlan on Catalyst 6509 w CatOS version 7 2 2 auxiliaryvlan on Catalyst 6509 w CatOS version 6 3 7 auxiliaryvlan on Catalyst 6509 w CatOS version 5 5 15 auxiliaryvlan on Catalyst 6509 w CatOS version 5 5 7a auxiliaryvlan on Catalyst 6509 w CatOS version 5 5 3a auxiliaryvlan on Catalyst 4000 w CatOS version 7 2 2 auxiliaryvlan on Catalyst 4000 w CatOS version 6 3 3 auxiliaryvlan on Catalyst 4000 w CatOS version 5 5 15 auxiliaryvlan on Catalyst 4000 w CatOS version 5 5 7a voice vlan on Catalyst 3524 with IOS version 12 0 5 WC11 voice vlan on Catalyst 3550 with IOS version 12 1 22 EA1 voice vlan on Catalyst 3560 with IOS version 12 1x Furthermore Avaya IP phones have been deployed on a broader range of CatOS and IOS platforms by various Avaya customers also with no known issues to date Therefore auxiliaryvlan voice vlan and explicit 802 1Q trunking are all viable options when a dual VLAN environment is required see Appendix A It is left to the user to choose the method keeping in mind that auxiliaryvlan and voice vlan are Cisco propri
29. packet not just the IP payload but the entire IP packet is fragmented and carried inside multiple ATM cells A 200 byte G 711 IP packet would require five ATM cells 25 octets of ATM overhead whereas a 60 byte G 729 IP packet would only require two ATM cells 10 octets of ATM overhead Refer to Appendix C for information regarding RTP header compression Keep in mind however that the same precautions apply to RTP header compression as to QoS see the next section on CoS and QoS The router could pay a significant processor penalty if the compression is done in software Inter LATA typically interstate Frame Relay is also affected by this ATM phenomenon This is because most carriers ATT Verizon Sprint convert Frame Relay to ATM for the long haul between the local central offices This is done through a process called frame relay to ATM network interworking and service interworking FRF 5 and FRF 8 In this process the Frame Relay header is translated to an ATM header and the Frame Relay payload 1s transferred to an ATM cell Since the Frame Relay payload can be a variable size but the ATM payload 1s a fixed size a single Frame Relay frame can be converted to multiple ATM cells for the long haul Therefore it is beneficial to limit the size of the voice packet even when the WAN link is Frame Relay 2 3 CoS and QoS General The term Class of Service refers to mechanisms that mark traffic in such a way that the traffic can be
30. priority device IP telephone should tag with VID 0 and the desired priority The low priority device PC would not tag at all No tag at all is the same as priority 0 default 2 Multi VLAN Ethernet switch port A multi VLAN port has a single port native VLAN and one or more additional tagged VLANs with each VLAN pertaining to a different IP subnet In general do not configure multiple VLANs on a port with only one device attached to it unless that device is another Ethernet switch across a trunk link For the attached device that belongs on the port native VLAN follow the points given for rule 1 above Clear frames untagged frames are forwarded on the port native VLAN by default An attached device that belongs on any of the tagged VLANs must tag with that VID and the desired priority The most common VolP scenario for a multi VLAN port is an IP telephone with a PC attached where the phone and PC are on different VLANs In this case the PC would send clear frames and the IP telephone should tag with the designated VID and desired priority As stated previously an Ethernet switch must be capable of interpreting the 802 1Q tag and many must be explicitly configured to receive it The use of VID 0 is a special case because it only specifies a priority and nota VLAN Avaya switches accept VID 0 without any special configuration but some Ethernet switches do not properly interpret VID 0 And some switches require trunkin
31. related to the information in section 3 4 heading C LAN Capacity and Recommendations This parameter only dictates when a warning is triggered and does not affect the total number of TCP sockets supported by the C LAN Although the recommended number of sockets on a C LAN may be less than 400 it is advisable in many cases to wait until 400 default value to trigger an alarm The parameter Receive Buffer TCP Window Size should be left at the default value of 8320 The default value should only be changed by AVAYA Services The Allow H 323 Endpoints and the Allow H 248 Endpoints fields are administered to allow or disallow registration of endpoints and gateways on the C LAN The Gatekeeper Priority parameter is used for Alternate Gatekeeper lists and 1s available when H 323 endpoints are allowed to register The lower the number the greater the priority data module Options are change display and list This form is used to assign an extension required for call processing to a C LAN board and to specify other parameters The Extension can be any valid extension in the dial plan and does not have to be a DID extension The Type is Ethernet The Port is the board slot appended with the number 17 ie 01A0517 The Link number can be any available number from the output of the display communication interface links command The Name is the previously defined node name ie c lan_80 KW Avaya IP Telephony Implementation Guide 3
32. scopes should have rotating varying gatekeeper lists so as to produce a uniform distribution of GRQs at boot up Most DHCP servers facilitate this by permitting the option 176 string to be created per scope which is KW Avaya IP Telephony Implementation Guide 54 recommended Do not create a global option 176 string that would apply to every scope on a server resulting in only one gatekeeper list Note that this principle may also apply to multiple TFTP servers Branch Site The branch site is just slightly different in terms of the DHCP scopes but very different in terms of the failure scenario and other factors that affect the branch implementation The IP telephones at the branch site could access the same four C LANs shown above or there could be a different set of C LANs not shown for the branch IP phones In either case the DHCP scopes for v80 and v90 should have rotating lists as at the main site However in addition to the list of C LAN addresses the v80 and v90 scopes should also include the S8300 LSP address at the end of the list This is because the LSP can take over as the call server for the branch if the WAN link fails The LSP only accepts registrations when it is active so having the LSP in the list does not result in inadvertent registrations to the LSP Because an extended WAN link failure is possible the branch site should ideally have its own DHCP server It makes sense that if there is a redundant call server at t
33. set of values for both signaling and audio Appendix F gives examples of how the L3 values are used in conjunction with QoS on routers L2 and L3 prioritization on the C LAN requires the TN799DP board with firmware v5 or later Direct IP IP Audio shuffling and IP Audio Hairpinning within a network region and across different network regions are enabled and disabled on this form Direct IP IP audio permits calls between IP endpoints to shuffle directly to each other instead of speaking through the MedPro MR320 board or VoIP module Ifa feature that requires the media gateway such as conferencing is activated during the call the endpoints shuffle back to the MedPro MR320 board or VoIP module If the conference ends and only two parties remain the IP stations shuffle back to one another Hairpinning permits calls between IP endpoints to speak through the MedPro MR320 board or VoIP module but without any transcoding This is essentially a relay feature for IP endpoints that are not capable of redirecting their audio streams None of the Avaya IP telephones have this limitation Direct IP IP Audio and IP Audio Hairpinning are generally enabled unless there is an Avaya R300 or MultiVOIP gateway in this network region in which case hairpinning should be disabled Also for direct IP IP audio to function across different network regions an inter region codec set must be specified and the regions must be connected via the inter region connectivi
34. set port vlan binding mode lt mod prt gt sets the vlan binding mode for given port s show trunk lt mod port gt displays trunking and vlan binding info for all ports or given port s show vlan displays vlan configuration information Avaya SAT and IPSI Interfaces change ip interface lt slot gt sets the speed and duplex for an IP board list ethernet options displays administered speed and duplex for all IP boards get ethernet options lt slot gt compares administered vs actual speed and duplex for an IP board IPSI commands These commands are executed from the IPSI IPADMIN prompt set port negotiation 1 enable disable enables or disables IPSI control port port 1 speed duplex negotiation set port speed 1 100MB 10MB sets control port speed set port duplex 1 fulljhalf sets control port duplex show port 1 displays control port status and configuration show control stats displays control port statistics and errors KW Avaya IP Telephony Implementation Guide 70 Appendix F Sample QoS Configurations This appendix gives simple examples of configuring QoS on Cisco routers It is only meant to give the reader a starting point Consult Cisco s documentation for a full explanation of Cisco s QoS implementation This rudimentary network configuration is used as a reference point The objective is to assure high quality of service to VoIP applications across the congested WAN link C LAN other subnet other subnet
35. sides accept incoming calls on an IP trunk in bypass state So if side A detects the IP network recovery first and calls side B while B is still in bypass state side B accepts the call However the same is not true if side B is in out of service state The scenario for severe congestion is the same Q5 If the C LAN or S8300 on one end of the IP trunk fails does the IP trunk cover to a different C LAN or S8300 KW Avaya IP Telephony Implementation Guide 76 No the IP trunk has fixed termination points If one of the points fails the IP trunk goes out of service almost immediately at the local system where the failure occurred This is especially true for an S8300 because it is the call server and not just a call signaling board like the C LAN At the remote system the other end of the IP trunk the IP trunk eventually goes out of service as follows The IP trunk bypass feature puts the signaling group in bypass state unless the system is an S8300 media gateway The Maintenance Function either at the normal interval or triggered by a call attempt puts the signaling group out of service Depending on which of these occurs first the signaling group may go into bypass and then out of service or out of service directly A way to compensate for this type of outage is to administer multiple IP trunks signaling groups and trunk groups across multiple C LANs between the same systems Q6 What about a MedPro MR320 or VoIP module failure at eithe
36. size is no larger than 250 hosts each for the voice and data VLANs High broadcast levels are particularly disruptive to real time applications like VoIP Avaya media servers and gateways and IP telephones utilize very low amounts of broadcast traffic to operate Therefore a subnet VLAN with only these Avaya hosts has a very low broadcast level There are two cases where Avaya hosts can be subjected to high levels of broadcasts 1 Avaya hosts and other broadcast intensive hosts share a subnet VLAN and 2 broadcast intensive PCs are attached to Avaya IP phones Case 1 is one of the reasons for the recommendation to use separate voice subnets VLANs Case 2 is sometimes unavoidable and the result is that broadcasts used by the PC must pass through the phone even if the phone and PC are on different VLANs For this reason Avaya IP phones are designed to be very resilient against broadcasts with lab tests showing the phones operating satisfactorily even with 3 000 to 10 000 broadcasts per second depending on the model Nevertheless to provide acceptable user experience and audio quality high broadcast environments are very strongly discouraged The recommended maximum broadcast rate is 500 per second and the absolute maximum is 1000 per second If VoIP hosts must share a subnet with non VolIP hosts not recommended they should be placed on a subnet VLAN of 250 hosts or less with as low a broadcast rate as possible Use 100M links take measures not t
37. supports 64 uncompressed audio streams G 711 codec or 32 compressed audio streams G 729 codec or any combination using the following formula uncompressed streams 2 compressed streams 64 The MR320 supports up to 320 audio streams depending on licensing and configuration Per the RTP standard each audio stream requires an even numbered UDP port for the RTP audio and the subsequent odd numbered UDP port for the RTCP control exchange Therefore to support X audio streams the UDP port range must contain 2X consecutive ports beginning with an even port and ending with an odd port Since the absolute maximum value for X is 320 MR320 board the largest required UDP port range 1s 640 Duplicated Media Resource 320 MR320 boards need 320x4 UDP ports or 1280 ports In this case a port range of 50000 to 51279 can be administered The DiffServ DSCP and 802 1p Q parameters are the L3 and L2 priority values for call signaling from C LANs in this network region and audio from MedPros MR320s in this network region The L2 values are only applied to boards that have L2 tagging enabled via the ip interface form The reason for the two forms is that L2 tagging and VID can vary per board across a network region but the priority values are typically uniform throughout the region Ideally two different sets of L2 L3 values should be specified for signaling and audio However for practical purposes in many applications it is common to use the same
38. t reach the far end because of the outage The status signaling group command at the S8700 shows the signaling group in bypass state The same command at the S8300 shows the signaling group in far end bypass state In this condition both sides use the fallback TDM trunk until the 8700 puts the signaling group back into service Q3 As a follow up to the previous question what are the effects of the two sides not detecting the outage at exactly the same time Both sides accept incoming calls on TDM trunks regardless of the state of IP trunks So if side A detects an IP network outage and calls side B via the TDM trunk instead of the IP trunk side B accepts the call Side B continues to attempt using the IP trunk until it detects the outage at which time it utilizes the TDM trunk for its outbound calls In the case of severe congestion side A detects the congestion first goes to bypass state and starts using the TDM trunk This causes side B to go to far end bypass state and also use the TDM trunk Eventually side B detects the congestion and goes to bypass state as well unless the system is an S8300 media gateway Q4 When the IP network recovers after an outage or severe congestion do both sides discover this at the same time and start sending calls over the IP trunk at the same time If not what are the effects No as with detecting the failure detecting the recovery is also independent But this is usually not a problem because both
39. the far end gatekeeper is associated That is the far end gatekeeper 1s treated as if it were an endpoint in the locally defined network region specified in this field RRQ Required y if the signaling group is for a G150 R300 or MultiVOIP gateway This requires the gateway to send a RAS Registration Request to bring the signaling group into service KW Avaya IP Telephony Implementation Guide 41 Media Encryption New to Avaya Communication Manager 2 1 This parameter permits media encryption between the two Avaya systems joined by this IP trunk Selecting y invokes a passphrase and both ends of the IP trunk must have the identical passphrase This facilitates a key exchange between the systems which makes media encryption possible between endpoints on the two systems as long as the ip codec set forms on both systems are configured with matching encryption options In other words enabling encryption on the ip codec set form permits encryption within a system Media encryption between two systems is possible when they have compatible codec sets and encryption options and are connected by an IP trunk with this feature enabled DTMF over IP See the section below for the system parameters ip options form Calls Share IP Signaling Connection y if the far end is an Avaya device n if it is another vendor s device y means that a single H 225 signaling connection is used for all trunk members all c
40. the H 245 UserInputIndication message is used to pass the digits Otherwise the Keypad Information Element of an H 225 Q 931 INFO message 1s used to pass the digits ttp payload The digits represented by the tones are sent via the RTP payload format specified in RFC 2833 This is required by SIP but also applicable to H 323 The last two options require the MedPro MR320 board and VoIP media module to detect the tones and remove them from the outgoing audio stream Then a message is sent to the call server for each digit to be sent out of band or a separate RTP packet with the specified payload format is created for each digit KW Avaya IP Telephony Implementation Guide 44 SAT Troubleshooting Commands The following table lists some common SAT troubleshooting commands status station lt ext gt list trace station lt ext gt list trace ras ip stations lt ext gt status signaling group lt group gt status trunk lt group gt or lt group gt lt member gt status ip board lt slot gt status clan port lt slot 17 gt ie 01a0517 status clan usage status media processor all board lt slot gt status ip network region lt gt list ethernet options get ethernet options lt slot gt ping and trace route Gives static view of a station s status multiple pages Gives real time view of a station s activities for tracing calls Traces a station s registration events GRQ GCF RRQ RCF
41. trunk port that has only one VLAN while Cisco removes excess VLANs from a trunk port that has all VLANs Either method achieves the desired objective KW Avaya IP Telephony Implementation Guide 57 which is to have only two VLANs configured on a trunk port connected to an IP phone so that broadcasts from non essential VLANs are not permitted to bog down the link to the IP phone VLAN Binding Feature P330 C360 On the Avaya P330 C360 additional VLANs are added to a port using the VLAN binding feature The port may be a trunk port 802 1Q tagging enabled or an access port no 802 1Q tagging The port does not need to be a trunk to forward multiple VLANs and for one application connecting to an Avaya IP phone it must not be a trunk ie do not issue the set trunk command The following steps enable VLAN binding 1 Verify that the port is configured with the desired port native VLAN 2 Add additional VLANs with one of the following vlan binding mode options Static option set port vlan binding mode lt mod port gt static Put the port in bind to static mode set port static vlan lt mod port gt lt vid gt Statically add another VLAN in addition to the port native VLAN Configured option set vlan lt id gt Add a VLAN to the configured VLAN list Type show vlan to see entire list set port vlan binding mode lt mod port gt bind to Apply the configured VLANs to the port and permit configured only those VLANs bind to
42. 1 0 0 0 0 255 dscp 46 right router Access list 101 permits any IP traffic that is marked with DSCP 46 between the two VoIP subnets There is an implicit deny any at the end of this access list class map match any VoIP create a class map called VoIP match access group 101 only packets matching access list 101 are in the class VoIP this is more restrictive than matching any packet with DSCP 46 the remainder of the configurations is identical to example 2 policy map voipQoS create a policy map called voipQoS class VoIP give strict priority to packets in the VoIP class on up to 816k priority 816 of this WAN link class class default put everything else in the default class and transmit it out the default fair queue queue in a weighted fair queue fashion random detect dscp based if the default queue starts to get full randomly discard packets in this queue based on DSCP lower values get discarded first interface Serial0 description T1 ip address 172 16 0 1 service policy output voipQoS apply the voipQoS policy outbound on this interface If any of the endpoints were incapable of DSCP marking the dscp 46 could be removed from access list 101 Then any traffic between the two VoIP subnets regardless of the marking would be in the class voip KW Avaya IP Telephony Implementation Guide 74 Appendix G IP Trunk Bypass TDM Fallback Q amp A Q1 How does the IP trunk bypass aka TDM fallback feature work and how should the parame
43. 10K BHCC Provisioning for VoIP must include the Layer 2 overhead which includes preambles headers flags CRCs and ATM cell padding The amount of overhead per VoIP call includes e Ethernet adds a 18 byte header plus a 4 byte CRC plus an optional 4 byte 802 1Q Tag plus a 8 byte preamble for a total of up to 34 bytes per packet Point to Point Protocol PPP adds 12 bytes of layer 2 overhead per packet Multilink PPP adds 13 bytes per packet Frame Relay adds 6 or 7 bytes per packet ATM adds varying amounts of overhead depending on cell padding IPSI encryption adds up to 23 bytes AES for the encryption header and padding in addition to Layer 2 overhead IPSI bandwidth calculations should include the additional overhead on a per packet basis depending on the type of WAN link For example for a busy hour call completion rate of 5K calls moderate general business traffic rate the L2 overhead for a PPP link would be 61 PPS X 12 bytes packet or 6 3 Kbps for additional PPP L2 overhead for a minimum of 58 1 Kbps Encryption would add 23 additional bytes per packet or and additional 11 2 Kbps for a total of 69 3 Kbps A general rule of thumb for IPSI Control traffic bandwidth allocation is to add an additional 64Kbps of signaling bandwidth to the minimum required bandwidth in order to manage peak burst traffic loads and either round up or down to nearest DSO Using the previous example of 5K busy hour calls using encrypted PPP links to contr
44. 20 protocols and ports C LAN and MedPro MR320 Network Placement Place the C LAN and MedPro MR320 boards on highly reliable subnets as close as possible to the majority of IP endpoints 1e IP phones and softphones Keep in mind that both call signaling and audio from all IP endpoints require these boards Therefore it may not be good practice to place these boards on a subnet containing many enterprise resources such as a server farm where there is heavy traffic both on the subnet and on the uplink s to the subnet On the other hand a server farm is typically where the most reliable and redundant network resources are deployed A thorough understanding of the network and network traffic is required to ultimately determine the best placement of these critical boards C LAN and MedPro MR320 Speed Duplex Use the SAT ip interface form to configure the speed and duplex for the C LAN and MedPro MR320 boards It should be standard procedure to properly set the speed and duplex on all C LAN and MedPro MR320 boards and to configure the associated Ethernet switch ports accordingly This results in much better system stability and audio quality than if the boards and Ethernet switch ports are left to auto negotiate See section 2 1 under the Speed Duplex heading KW Avaya IP Telephony Implementation Guide 33 The default speed duplex setting on the MedPro MR320 board is auto negotiate The default speed duplex setting on the
45. 7 ip codec set Options are change display and list This form is used to define the codec sets that are referenced by other IP telephony forms Up to 7 codec sets may be defined with 5 codecs listed in order of preference in each set G 711 uncompressed and G 729 compressed are the recommended codecs for LAN and WAN respectively No silence suppression and 20 ms voice packets are also recommended A word of caution CM allows for the administration of the G 726A codec type but it is only available on the MR320 TN2602 The TN2302 does not support G 726A Note about silence suppression Although silence suppression conserves bandwidth by not transmitting audio packets during periods of silence its use typically results in audio clipping which most users consider unacceptable The G 729B codec may be a better alternative to silence suppression Rather than not transmitting during silence this codec transmits silence in a condensed format that requires less bandwidth The audio quality of G 729B is still noticeably inferior to G 729 Note about voice packet size Audio is encoded in increments called frames with the typical frame size being 10ms The packet size or number of frames per Larger packet size less bandwidth Smaller packet size more bandwidth Larger packet size low loss high jitter network packet is a measure of how much audio is sent in each Smaller packet size high loss low jitter IP packet Experi
46. AN but phone is already on it Using this method the phone applies the L2 priority values for audio and signaling as administered on the ip network region form for the phone s region Using the recommended DHCP option 176 method the phone applies the L2 priority values received from DHCP The Emergency Location Extension is part of the E911 features of Communication Manager and is not within the scope of this document station Options are add change display and list This form is used to define stations To specify an IP station the Type must be an IP model The Port is automatically set to X for an IP phone when the station is first added This is changed to S an automatically assigned internal port number when the phone registers with the call server The IP Softphone inquiry is regarding whether or not a softphone is permitted to take over the extension This field applies to non IP stations as well as an IP softphone can take over an analog or DCP extension and emulate that set type Survivable GK Node Name provides an option for the station to fail over to an Avaya G150 Media Gateway or a MultiTech MultiVOIP gateway when no other gatekeeper is available Direct IP IP Audio and IP Audio Hairpinning for the individual station is configured on page 2 of this form trunk group and signaling group Options are add change display and list These forms are used to define trunks including H 323 IP trunks This document
47. AVAYA Avaya IP Telephony Implementation Guide Communication Manager 3 1 Avaya Labs ABSTRACT This document gives implementation guidelines for the Avaya MultiVantage Communications Applications Configurations and recommendations are given for various Avaya Media Servers and Gateways as well as Avaya 4600 Series IP Telephones This document also provides information on virtual local area networks VLANs and guidelines for configuring Avaya and Cisco networking equipment in VoIP applications The intent of this document is to provide training on IP telephony and to give guidelines for implementing Avaya solutions It is intended to supplement the product documentation not replace it This document covers the Avaya Communication Manager 2 2 through 3 1 and the Avaya 4600 Series IP Telephone 1 8 and later with limited information regarding previous and future versions External posting www avaya com May 2006 COMPAS ID 95180 Avaya IP Telephony Implementation Guide All information in this document is subject to change without notice Although the information is believed to be accurate it is provided without guarantee of complete accuracy and without warranty of any kind It is the user s responsibility to verify and test all information in this document Avaya shall not be liable for any adverse outcomes resulting from the application of this document the user accepts full responsibility The instructions and tests in this docu
48. Avaya Communication Solutions and Integration CSI group One essential function of this professional services group is to provide pre deployment network assessments to Avaya customers This assessment helps to prepare a customer s network for IP telephony and also gives critical network information to Avaya support groups that will later assist with implementation and troubleshooting Arrange for this essential service through an Avaya account team KW Avaya IP Telephony Implementation Guide 3 Avaya IP Telephony Implementation Guide Table of Contents 1 Introduction to VoIP and Avaya Products ccccccccccccccccccceeeeeeeeeeeeeeeeeeeeeeeeeeeeseeseeeeeeeeeeeesesaeseaaaaaaaeaaaaas 7 1 1 Servers Gateways Stations and Trunks Defined 00 cc cccccccesseeeseeeeeeeeeesseessssessssssssseeeaaas 7 VES acc senses cesses we ao sts SE eS GT Sante em 7 IGE WAYS E E E E E E E N E E AE 7 BAUM S PAA E E A E A A E 7 OI a i AIK ceases siete AA ssi arses ote ee ee te ee se terrae cet wa ae E OANA E OEN caine EE 7 1 2 Avaya Server Gateway and Trunk Architectures 0 00 cccccccccccsessescssscessssessssssssssessesssseeeeeeeees 8 Traditional DEFINITY System a ccrcaceteau tosctenvsccucrontcdoesnaeseaethaseaeenntasgudnasaeatnaie E AREE EE RS 8 IPenabled DEFINITY Syste M ce sacerctaneeeeieenecalseueeaasacieinacenceaneesaceananateusnaasecuasacenceaneesacaaneaniedseacsact 9 IMM BNE sec cts EEA EENE EAEE EE toa naa eaata ecu nin ec ce
49. H 248 MG to primary field indicates that the rule is disabled The failover to an LSP and recovery back to the primary server are covered in detail in the Avaya Communication Manager Network Region Configuration Guide at www avaya com system parameters ip options Options are change and display This form is used for miscellaneous IP settings IP Media Packet Performance Thresholds These parameters detailed in Appendix G are for the IP trunk bypass feature described in the section covering the signaling group form RTCP Monitor Server These are the VoIP Monitoring Manager server settings applied to all network regions unless specified otherwise in the ip network region form Automatic Trace Route on Link Failure This feature relates to the following links Port network control link between S87xx S8500 server and IPSI board KW Avaya IP Telephony Implementation Guide 43 H 248 media gateway control link between CLAN S8300 and media gateway IP trunk between near end system and far end system When this feature is enabled and Communication Manager detects a failure on one of these links Communication Manager launches a trace route from the source of the link to the destination of the link A failed trace route might indicate that the link failure was associated with a network fault whereas a successful trace route might indicate otherwise This feature should be disabled if ICMP is blocked on the network so as
50. Office 58700 s8700 H 225 RAS amp m S i Q 931 signaling S s OO E he _ Iis 4 9700 with a 12 switch L2 switch Control iP iP i i ma ae a M IP Network backup CCMS 225 oN NEES ae Se o PN T PN a IP Net gt 9700 with A nM lee VoIP mod a C LAN a gae g350 with AAA MedPro VoIP mod nN i sh DD C LAN s i n Q Q es S oe g700 with RTP VoIP mod SCC MCC MCC Enterprise IP IP DCP mod IP Network ae Analog mod CCMS and bearer gt i i a T1 E1 mod or ATM PNC gt gt ii DCP Analog S Q Figure 7 Multi Connect with remote G700 G350 G250s Remote gateways and stations are controlled by the S87xx servers via the C LAN boards The remote 8300 is in local survivable processor LSP mode to take over as call server if connectivity to the S87xx servers 1s lost KW Avaya IP Telephony Implementation Guide 12 IP Connect with Remote G700 G350 G250 Gateways Medium Large Enterprise 8700 58700 Enterprise IP Network CCMS over TCP IP Remote Office backup H 225 ae g700 with s8300 LSP VoIP mod are g700 with VoIP mod g350 with VoIP mod g 5 gt m j m La 1 mI aF RTP VolP mod IP IP DCP mod ae Analog mod a T1 E1 mod DCP Analog N Q Figure 8 P Connect with remote G 700 G350 G250s connectivity to the S87xx servers is lost KW Avaya IP Telephony Implementation
51. RO TPO and DSC P eoar E sean caceeteaseeneea Ronee etceceeet 28 Eo OO CLV CL A E I E ET E TE E E N SE A E E E E 28 3 2 wpe a OO S062530 Media Gate Way Siisera E E TE 29 GWP SOUC 360 LZS Wie iere N N E 29 G700 Media Gateway Processor MGP saseicsasdiczaxechcarelarccseatenndeveedepnsstttavaceciensdatiiandcadiemiatexas 29 C700 S02 1 O an DSE Poa aes eietatasari tonneau aes cran casei ealie wn iaied tant taled 30 77 00 tii Octaplan Stack VS Standalone ssiri N 30 GSO Medid GUE WIV e rer N ENA ENA NRN 31 COMAE WN A aa E A EA A S 31 General Guidelines Related to GatewayS cccccccccccccccessseccsseessssssssssseeessasasaseaasasaasaeaaaaaaaaaes 32 3 4 G650 G600 MCC1 and SCC1 Gateways Port Networks ccccccccccccccceceeeeeeeeeeeeeeeeeeeeeeeees 32 C LAN Capacity and Recommendations ccccccccccccccccceeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeseseeeeseess 32 C LAN and MedPro MR320 Protocols and Ports cccccccccccccceeeeeseeceeeeceeeeeeeeeeeeeeeeeeeeeeeeeeees 33 C LAN and MedPro MR320 Network Placement ccccccccccccccssseeeeeeeeeeeeeeeeeeeeeeeeseeseeeeeeeeas 33 C LAN and MedPro MR320 Speed Duplex orineriinsean aise cso eee 33 C LAN and MedPro MR320 802 1p Q and DSCP c cccccccccccccecectececeestecatecstecteesetecececsteases 34 MR320 Capabilities and MR320 Bearer Duplication ccccccesesssssseessessesssseesssssesssseeeeens 34 Extreme Measures for MedPro and Other IP Boards on Cisco Swit
52. S The L3 router commands are practically the same as on the X330WAN and P330R routers using commands similar to Cisco s IOS The G350 utilizes both command sets in a single CLI The G350 can have several physical and logical interfaces and there is no single inband interface as on the G700 s P330 switch component One of the G350 s IP interfaces must be designated the primary management interface PMI There is a default designation but it can be changed by inserting the command pmi under the desired interface The PMI among other things is the interface used by the MGP and internal VoIP module This means that H 248 media gateway signaling and RTP audio are sourced by and terminated on the PMI When an S8300 media server is inserted into a G350 one of the VLANs on the G350 must be designated the ICC VLAN This is done by inserting the command icc vlan under the desired VLAN interface The remaining G350 G700 similarities and differences are as follows Refer to the G700 sections above for more details on each subject Like the G700 the G350 needs at least a default route to route off of its connected VLANs subnets But there is a single default route for the whole unit as opposed to a route for the P330 component and a route for the MGP component This is configured using the ip route command The G350 and G700 L2 switch configurations Spanning Tree VLANs trunking speed duplex are very similar The G350 and G700 802 1
53. The Avaya S8300 Media Server in internal call controller ICC mode is the call server The 88300 is a Linux platform similar to the 8700 but in a compact form factor that fits into these gateways The 8300 is not front ended by C LANs it terminates the call signaling natively KW Avaya IP Telephony Implementation Guide With P Connect the traditional port networks MCCI and SCC1 are replaced with new 19 inch rack mountable Avaya G650 or G600 Media Gateways All G650 G600s require IPSI boards no more Center Stage or ATM PNC Everything is done on the enterprise IP network no more control IP network G650 G600 media gateways still use C LAN and MedPro MR320 boards as well as the other traditional port boards used in the MCC1 and SCC1 Small Medium Enterprise i H 225 Se E g700with IP IP L i 58300 ICC VolP mod fy m E g700 with H 248 media viele mod gateway control gt g350 with VolP mod Ent IP Net ais i ee ap RTP VolP mod IP IP DCP mod p Analog mod a T1 E1 mod DCP Analog Es m Tie Figure 6 S8300 G700 G350 G250 architecture Multi Connect with Remote G700 G350 G250 Gateways Medium Large Enterprise Remote
54. a Gateway H 248 term Port Network Avaya term A gateway requires a call server to function and some common Avaya server gateway architectures are illustrated later Stations There are several technical terms for what most would call a telephone and some that are generally used throughout the industry are listed below Endpoint H 323 general term that includes IP phones and other endpoints Terminal H 323 specific term to mean primarily IP phones also an Avaya term Station Avaya term and possibly a generic term Set Avaya term and possibly a generic term Avaya gateways have port boards or media modules that terminate various types of stations Trunks Trunks connect independent telephony systems together such as PBX to PBX or PBX to public switch or public switch to public switch In traditional telephony there are various types of circuit switched trunks using various protocols to signal across these trunks IP telephony introduces another trunk type the IP trunk Like circuit switched trunks IP trunks connect independent telephony systems together but the medium is an IP network and the upper layer protocol suite 1s H 323 Avaya gateways have port boards or media modules that terminate various types of trunks including IP trunks KW Avaya IP Telephony Implementation Guide 7 1 2 Avaya Server Gateway and Trunk Architectures The following figures illustrate some common Avaya server g
55. a TCP with port 2945 1039 if encrypted on the media gateway controller side CCMS is an Avaya proprietary protocol for port network control same as media gateway control It is transported via TCP with port 5010 on the port network IPSI board side KW Avaya IP Telephony Implementation Guide 15 2 IP Network Guidelines This section gives general guidelines and addresses several issues related to IP networks LAN WAN and device configurations 2 1 General Guidelines Because of the time sensitive nature of VoIP applications VoIP should be implemented on an entirely switched network Ethernet collisions a major contributor to delay and jitter are virtually eliminated on switched networks Additionally VoIP endpoints should be placed on separate subnets or VLANs separated from other non VolIP hosts with preferably no more than 500 hosts per VLAN This provides a cleaner design where VoIP hosts are not subjected to broadcasts from other hosts and where troubleshooting is simplified This also provides a routed boundary between the VoIP segments and the rest of the enterprise network where restrictions can be placed to prevent unwanted traffic from crossing the boundary When a PC is attached to an IP telephone even if they are on separate VLANs both PC and phone traffic including broadcasts traverse the same uplink to the Ethernet switch In such a case the uplink should be a 100M link and the recommended subnet VLAN
56. actual speed duplex settings against configured settings for all boards and individual boards respectively With each new system or IP board installation one standard procedure should be to apply matching speed duplex settings to each IP board and its corresponding Ethernet switch port node names ip Options are change and display This form is used to define arbitrary names and associate an IP address with each name For example the name c lan_80 could be defined to describe a C LAN board on the 80 subnet with address 192 168 80 10 and the name medpro_ 80 could be defined to describe a MedPro board on the 80 subnet with address 192 168 80 11 KW Avaya IP Telephony Implementation Guide 36 ip interface Options are change display and list This form is used to configure individual IP boards The first step is to associate a board Type and Slot to a previously defined Node Name and to give the board a Subnet Mask and default Gateway and assign it to a Network Region For example the board type C LAN in slot 01A05 can be associated with the node name c lan_ 80 defined earlier This assigns the IP address 192 168 80 10 to the C LAN board in slot 01A05 Then the board can be given the mask 255 255 255 0 with default gateway 192 168 80 254 The board can also be assigned to network region 1 802 1p Q tagging for an IP board 1s also enabled or disabled on this form A number including 0 in the VLAN field indicates the VID
57. aeine 10 S8500 Media SClV EP sisi dcccdsswaciacnysoucsesh vevdenasedacucewandoasessesWeueedwhsavenssedddawssdueseaseckianvebuedeswaddaenesdads 10 IP Connect lt 0cccccsconansdencdseccanwiexensnniasensecuagasbiedainteamnces due neiansaebendGasbandaiaboannneuauenniaxesetendenebasdkeswnsneness 11 S8300 0700 0330 G2 30 sessar ARR REE E RRi 11 Multi Connect with Remote G700 G350 G250 Gateways ccccccsessssssssssessssssssssssssessssseaaes 12 IP Connect with Remote G700 G350 G250 Gateways ccccccccccssssssssssssssssssssseeeeeeeeeeeeeeeeeeees 13 TD MOE A steerer cet caterers ec esses cos serio cuts sc eon pvc arses dnote mec i ots omnis osc anata 14 1 3 VoIP Protocols and Ports vs4 nno2asseceesaraucasutua ecu wuanancncbasausaonad onan neasaensaaeaeaaoeae anon ummawcunnadauanebnncieats 15 2 IP Network Guidelines ccc ccc cccceccscceccecceccaccaccasccsccscceccecascescescesccscescescescescessesccacceacascescascascess 16 2 1 General Guidelines ccc ccccecceccecceccscecceccescescesceecceccecceceeceeccecseccecceccescescescscscuscascescescueseuscs 16 Ethernet Swit ches c cccccscosccsccscceccecceccsccsccsccsccsecescascasccceccesceccessescescessesceseceseaecasceccescess 16 BS NS a EA A A EA A A A AAN 17 2 2 Bandwidth Considerations ccceccecceccccecccscacccecceccecceccecceccescescescuscuecuecescusseeceeceeseeseeseeseuscs 18 Calculation vanccocsieseateatadasssanaae nsesxaieswhoustemsabnias
58. ainaietaananameaasendesnase EEEN EA 18 Ethernet Overhead 0 cccccccsccsccsccecceccecceccascscecccsccsccsccecescescssccsccescaccecescescesccecceccascascescess 19 WAN Overhead cccccsccsccecceccecccccacceccoscesccsccsccscescescsscsscesececceataccsceccescscscescescescescesceecs 19 Lo Fragmentation MTU goer omctsestnesnerceren nese attest nadine se ne pees ENE ETTA E RAEN 19 DB oO a0 Foe otec sce coisererieesspnceonciose AEE EEPE AEE EEPE AE 20 PA PO CO TANGO e PEET AA EA ENIA TEREA 20 CE e E E aniseed ceacesisavedentaalesnnuBieucnaaisiewaneedues 20 C A A A A 20 O Fo aren ne ee E A GN EAT EOE E OEA EOE E RE AE O eee 21 R l tor 302 1P OE Ah seieren aa Eeen ER A E E E EE A EE Na a 21 E e E E EE EEEE EEE EE ETE E EAEE AE AEE T 23 OOS Of an Ethernet S Wit Ds ciscasccsdncnnscustonseeucaramacesianaceadneumasudioavasoadenuessrsiaeediedesvassusveusspudoneeasisies 24 RO OT A OS acca SIA E E E AE E E E E AE T 24 OC SS E E E A E T sen 25 Trafe Shaping on Frame Relay LINKS 3p iic0ssccscenaactanasdenavecteaasncedgandeacsactandedeaadeectacenteeteanseostes 26 3 Guidelines for Avaya Servers and Gateways cccccccccccccccssseeeeeeeeeeeeeeeeeeeeeeeeeeeeeseeeescuseasaasseaeeeaeaeeaeaaas 27 KW Avaya IP Telephony Implementation Guide 4 4 KW De wep Or OKT ODIO CE VC A E E a ce cid E E to ce ede cate een ah tee ede ala oes 27 SS IRIX SIO0 Speed IB I0 0 Up eae rtp en reete ements cre ttre nce E eee nee ee ee 27 Sox SS NO
59. al to configure the speed and duplex correctly on the server interfaces used to communicate with the IPSI boards A speed duplex mismatch between these interfaces and the Ethernet switch causes severe call processing problems KW Avaya IP Telephony Implementation Guide 27 The web admin screen has a pull down menu for the various speed duplex settings This pull down menu does not indicate the current configuration but only the available options A current speed description next to this pull down menu indicates the current speed and duplex but it does not indicate whether these settings were manually configured or reached via auto negotiation Follow these steps to properly configure the speed and duplex Start with the server and the Ethernet switch port set to auto negotiate default The server should show Current speed 100 Megabit full duplex on the web admin screen and the Ethernet switch port should show that the negotiated speed duplex is 100 full When in doubt always return to this base state On the server manually configure the interface to 100 full With the Ethernet switch port still at auto negotiate it should now show that the negotiated speed duplex is 100 half This is expected Manually configure the Ethernet switch port to 100 full After a screen refresh the server should still show the current speed to be 100 full Both sides are now optimally configured for 100 full operation If either
60. alls and n means that each trunk member each call uses a separate signaling connection The G150 R300 and MultiVOIP gateway require this to be set to y Bypass if IP Threshold Exceeded Part of a feature commonly referred to as TDM fallback or IP trunk bypass This parameter has to do with whether or not a TDM fallback trunk is utilized when the IP network fails or performs poorly between the near end and far end gatekeepers The thresholds for this fail over are configured in the system parameters ip options form as described in Appendix G Appendix G is a Q amp A discussion on the IP trunk bypass feature and associated issues related to IP trunks Direct IP IP Audio Connections y typically same as with endpoints IP Audio Hairpinning y unless G150s R300s or MultiVOIP gateways can talk across the trunk The LRQ Required parameter allows IP trunk availability to be determined on a per call basis When this option is enabled a RAS Location Request LRQ message is sent to the far end gatekeeper prior to each call over the IP trunk The far end gatekeeper responds with a RAS Location Confirm LCF message and the call proceeds The absence of an LCF from the far end gatekeeper indicates that the call cannot proceed If this occurs and the near end gatekeeper is configured with the necessary route pattern the next preferred trunk in the route pattern is used for that call as follows S
61. an 10 v10 configured set port vlan binding mode 1 1 bind to Port bound to configured VLANs 1 and 10 configured set trunk 1 1 dotlq Port connected to Cisco router is an 802 1Q trunk port set port spantree disable 1 1 Spanning Tree disabled at the port level set port vlan 10 Port native VLAN changed to 10 on this port set port spantree disable 1 2 Spanning Tree disabled at the port level set port level 6 Port L2 802 1p priority set to 6 set port vlan 10 1 3 set port spantree disable 1 3 set port level 1 3 6 set port vlan binding mode 1 5 bind to Bound to configured VLANs but not a trunk port configured set port spantree disable 1 5 Spanning Tree disabled at the port level If the P330 C360 switch were a Cisco CatOS switch instead All ports have port native VLAN 1 by default set port host First invoke this command on all user ports Cisco switches do not tag the native VLAN but the router expects a set vlan 1005 1 1 tag on vl so the native VLAN is changed to some unused VLAN set trunk 1 1 on dotlq Port connected to Cisco router is an 802 1Q trunk port clear trunk 1 1 2 9 11 1004 Unnecessary VLANs removed 1 10 and 1005 remain set vlan 10 Port native VLAN changed to 10 on this port set port qos 4 cos 6 Port L2 802 1p priority set to 6 set vlan 10 1 3 set port qos 1 3 cos 6 Auxiliaryvlan is the more common method instead of explicit trunking set port auxiliaryvlan 1 5 10 v10 is the auxiliaryvlan on
62. and it means that tagging is enabled on the board with that VID Although most implementations where tagging is enabled should use VID 0 other VIDs are permitted as well The letter n in this column means that tagging 1s disabled on the board and a blank means that tagging is not supported on the board To properly enable L2 tagging on the C LAN and MedPro MR320 boards follow the instructions in section 2 3 under the heading Rules for 802 1p Q Tagging The speed and duplex settings for an IP board are configured on this form under the Ethernet Options heading The TN2602AP MR320 board has a VOIP Channels parameter to indicate how many channels are active on the board While this parameter is configurable it is restricted by licensing The initial licensing options are to purchase a number of boards with 80 channels each and a number of boards with 320 channels each The 2602AP MR320 board can be administered for Critical Reliable Bearer The Shared Virtual Address is the virtual ip address owned by the currently active MR320 and must be administered in the same subnet as the real 1p addresses The duplicated MR320s also share a virtual MAC address that is automatically assigned by one of four virtual MAC tables Each Virtual MAC table contains 64 cached AVAYA owned MAC addresses and each table can be displayed with display virtual mac table SAT command The C LAN board parameter Number of CLAN Sockets Before Warning This is
63. at the Ethernet level without requiring IP addresses Cisco devices identify themselves to other Cisco devices using CDP packets that contain device and port specific information CDP packets can be captured and decoded using protocol analyzers that support CDP With the appropriate devices and OS versions the CDP packets contain information specific to VoIP and other real time applications 1 p 2 22 Using CDP the Catalyst sends the Cisco IP phone an auxiliaryvlan ID if auxiliaryvlan is enabled and the phone tags its frames to be forwarded on that VLAN The auxiliaryvlan is the second Cisco proprietary mechanism and it must be enabled on the port that connects to the IP phone Itis VLAN 200 by default or can be arbitrarily assigned as any number between and 1000 According to Cisco s documentation the auxiliaryvlan is just another 802 1Q VLAN The only difference is the proprietary method of assigning it to a Cisco IP phone The port with the auxiliaryvlan also has a port native VLAN VLAN 1 by default or any arbitrarily assigned VLAN This implies that the port is an 802 1Q trunk port with two VLANs and can accept 802 1p Q tagged frames This is similar to the VLAN binding feature on the Avaya P330 v3 2 8 and later 1 p 2 22 2 23 The information passed from the Cisco phone to the Catalyst is not of concern The phone communicates its specific power requirements to the Catalyst and the phone can also trigger the Catalyst to se
64. ateway architectures in succession from established to most recent technologies Also included in the diagrams are the protocols used to communicate between the various devices Traditional DEFINITY System Adjunct Location Medium Large Enterprise Main Location oat CCMS from processor cals za to port boards across backplane te ik Oo oO eae Jf A E EPN T TDM bus TDM bus Sp iT DCP SCC SN MCC DCP amp CCMS and bearer Center Stage amp res Analog k over TDM or ATM or ATM PNC F nalog KW Figure 1 Traditional DEFINITY System architecture The single SCC1 and multi carrier cabinets MCC1 are called port networks Avaya term or media gateways VoIP term They house port boards which among other things terminate stations and trunks These port boards are not the focus of this document The DEFINITY call servers are the processor boards inserted into the processor port network PPN The other cabinets without processors are called expansion port networks EPN and are controlled by the DEFINITY servers in the PPN The port networks are connected together via a port network connectivity PNC solution which can be TDM based Center Stage PNC or ATM based ATM PNC Both bearer audio and port network control are carried across the PNC solutions Control Channel Message Set CCMS is the Avaya proprietary protocol
65. ation MTU lowering the MTU L3 fragmentation is much less efficient than LFI L2 fragmentation because it incurs additional L3 overhead as well as additional L2 overhead Lowering the MTU is generally not advisable and may not provide any added value because it adds more traffic to the WAN link than LFI The added congestion resulting from the increase in traffic may effectively negate any benefit gained from reducing serialization delay One should have a thorough understanding of the traffic traversing the WAN link before changing the MTU Because of all these configuration variables properly implementing QoS on a router is no trivial task However it is on the router where QoS is needed most because most WAN circuits terminate on routers Appendix F contains examples of implementing QoS on Cisco routers This appendix does not contain configurations for all the issues discussed in this document but it gives the reader a place to start QoS Guidelines There is no all inclusive set of rules regarding the implementation of QoS because all networks and their traffic characteristics are unique It is good practice to baseline the VoIP response ie voice quality on a network without QoS and then apply QoS as necessary Conversely it 1s very bad practice to enable multiple QoS features simultaneously not knowing what effects if any each feature 1s introducing If voice quality is acceptable without QoS then the simplest design may be a wi
66. ause it is a last resort measure On rare occasions a MedPro board s Cisco switch port may flap up and down continuously This is manifested by bridge join leave messages for CatOS based switches and interface up down messages for IOS based switches Sometimes this problem is caused by the backplane I O cable not being Cat5 compliant and Avaya Tier 3 support can determine whether or not this is the case Sometimes this problem is a compatibility issue between the MedPro and the Cisco switch After the instructions in section 2 1 headings Ethernet Switches and Speed Duplex have been followed if the Cisco switch port continues to flap up and down consider the options described in the next paragraph The Cisco white paper Troubleshooting Cisco Catalyst Switches to Network Interface Card NIC Compatibility Issues 4 p 6 describes the flapping problem mentioned above and offers a suggestion to KW Avaya IP Telephony Implementation Guide 35 adjust the jitter tolerance not related to audio jitter on Cisco switches The CatOS global command which is hidden is set option debounce enable disable to undo This command increases the jitter tolerance to 3 1 nsec from the 1 4 nsec default The IOS interface command is carrier delay 4 no carrier delay to undo This adjusts the carrier transition delay to 4 seconds If these commands do not correct or improve the flapping condition put the switch back to its original state and try
67. cess List Guidelines This appendix gives guidelines for configuring access lists to facilitate basic Avaya IP telephony functionality The ports used by the Avaya call server are fairly fixed and known The ports used by the endpoints are more variable and random As a result it is simpler to tailor access lists based on call server ports eisai ies or Protocol or Protocol The C LAN uses UDP port 1719 for endpoint registration RAS The C LAN uses TCP port 1720 for H 225 call signaling This is to facilitate IP trunking between two Avaya call servers and must be done for each IP trunk This is one way to facilitate audio streams between MedPros MR320s and endpoints UDP port range Any endpoint UDP any in 1p network region form UDP port range in Permit Any MedPro MR320 ip network region form Any endpoint UDP any Any MedPro MR320 This is another way to facilitate audio streams between MedPros MR320s and endpoints Any MedPro MR320 RTP RTCP Any endpoint fs Any endpoint RTP RTCP Any MedPro MR320 gt ns This is to facilitate audio streams between direct IP IP shuffled endpoints Permit Any endpoint UDP any Any endpoint UDP any RTP RTCP The R300 uses this default UDP port range for audio However the range is configurable RTP RTCP endpoint ie endpoint RTP RTCP Permit Any R300 UDP 1900 2075 Any R300 UDP 1900 2075 ee eee a N These are all services used by the IP telephone TFTP is toug
68. changed to 10 on this port Spanning Tree fast start feature Port native VLAN L2 802 1p priority set to 6 802 1Q trunk port Since most PCs do not understand the tag the PC s VLAN must be the native VLAN v1 is already the native but command shown VLANs 1 and 10 allowed on trunk Port is in trunk mode Spanning Tree fast start feature Simpler configuration on newer IOS switches ie 3550 3560 Access mode explicit trunking not required Configure the data VLAN Configure the voice VLAN This procedure applies regardless of the Eth switch used Initially placing the IP phone on VLAN 10 requires two DHCP scopes one for VLAN 1 and another for VLAN 10 Both scopes should have identical DHCP option 176 strings with one exception The VLAN 1 scope must have the LZAQVLAN parameter and the VLAN 10 scope should not The following strings apply to phone firmware 1 8 and beyond VLAN 1 MCIPADD addr1 addr2 JHTTPSRVR addr L2QVLAN 10 L2QAUD 6 L2QSIG 6 VLANTEST 0 VLAN 10 MCIPADD addr1 addr2 JHTTPSRVR addr L2 QAUD 6 L2QSIG 6 VLANTEST 0 Run the phone through its normal boot up sequence It obtains an IP address on VLAN 1 the port native VLAN When the phone receives the option 176 string above from the VLAN 1 scope it releases the VLAN 1 address and enters a second DHCP sequence with tagging enabled to obtain a VLAN 10 address After the phone is operational on VLAN 10 on subsequent reboots the phone re
69. ches cc csseeeeeeeeeees 35 IP Server Mmtetiace CIPS Board 5 sin sex eetieeseiissasSeachonesbatw uses sostaacdaapdancaheodektiaeueeash Goactadoveceonke 36 3 5 General IP Telephony Related Configurations SAT Forms cccccccccccccecceceeeeeeeeeeeeeeeeeees 36 1A 0 OU ONIONS ene an rr ee ere eee ere errr 36 Hode Names IP ienie ra ECL nnn ne ev NA nD ee ee ee 36 HD WNC AC erate Ares E ate ae niet net ee ae aera eee eee 37 CACAO CNG seco ncdenctacntincenidtanasietecceeeteaeaaat ace ERA 37 IP COde CSET al N E AT E AOT A ae line oleate tact Sh 38 FP IC PWV OUI TES ION nsnsi a e S DA E E TR E 38 D EG Dy OR SINAN a Sc E E css aca E a AN 40 SO A 41 trunk eroup and signaling group 26 ceecsesen ss toe sec caesar ai a i a ia a 4 WIC CASO ALC Wy Ay en aaa raw 0s acs waco eset a 43 systenm parameters Mo recovery rullen rir ennn ded AREE ATREA oes 43 systemi paraineters IP OPHONS ardan Eaa EE E 43 SAT Troubleshooting Commands serria A EATR 45 Guidelines for Avaya 4600 Series IP Telephones eneeeeeneenenennensnssssssssnssesnssnsssnsnnnnsnnennessssssssssss 46 2 N 0 e T E E E S A EE E E E EE E E E 46 Legacy Models vs Curent Mod l cerauni E aR 46 TTL G PHO pu Oi LIO rei A 47 DHC RLE c D O yee coe ne meme tne Nene entre parent rt ORO etn et cee enema ee onan eee een renee 48 Additional Script and Firmware Download Methods cccsssssessseeesseeseseseessssssesssssseeeees 48 TOOL UP DE GU CNC So anena a n a a
70. ction to software or for any other reason Since most QoS KW Avaya IP Telephony Implementation Guide 25 policies are implemented on WAN links the following very general points for Cisco routers are offered to increase the level of confidence that QoS remains in hardware Consult Cisco to be sure Newer hardware platforms are required 2600 3600 7200 and 7500 Newer interface modules WIC VIP etc are required Consult Cisco to determine which hardware revision is required for any given module Sufficient memory is required Device dependent Newer IOS is required 12 0 or later Several things should be examined whenever QoS 1s enabled on a network device First the processor level on the device should be examined and compared to levels before QoS was enabled It is likely that the level will have gone up but the increase should not be significant If it is significant then it is likely that the QoS process is being done by software The processor load must remain at a manageable level max 50 average 80 peak If the processor load is manageable the VoIP response should be examined to verify that it has improved under stressed conditions ie high congestion compared to performance before QoS was implemented There is no added value in leaving a particular QoS mechanism enabled if VoIP response has not improved under stressed conditions If VoIP response has improved then the other applications should be checke
71. d C360 Cisco CatOS set port vlan lt id gt lt mod port gt set vlan lt id gt lt mod port gt All clear Ethernet frames ones with no 802 1Q tag such as from a PC are forwarded on the port native VLAN This is true even if the Eth switch port 1s configured as an 802 1Q trunk or otherwise configured for multiple VLANs see VLAN binding heading below Configuring a Trunk A trunk port on an Eth switch is one that is capable of forwarding Ethernet frames on multiple VLANs via the mechanism of VLAN tagging IEEE 802 1Q specifies the standard method for VLAN tagging Cisco also uses a proprietary method called ISL Avaya products do not interoperate with ISL A trunk link is a connection between two devices across trunk ports This can be between a router and a switch between two switches or between a switch and an IP phone Some form of trunking or forwarding multiple VLANs must be enabled to permit the IP phone and the attached PC to be on separate VLANs The following commands enable trunking Avaya P330 and C360 Cisco CatOS set trunk lt mod port gt dotlq set trunk lt mod port gt nonegotiate dotlq By default only the port native VLAN is enabled on By default all VLANs 1 1005 are enabled on the the trunk port Another set of commands is required trunk port VLANs can be selectively removed with to specify other allowed VLANs the command clear trunk lt mod port gt lt vid gt Note that Avaya adds additional VLANs to a
72. d method was implemented as of Avaya Communication Manager 1 3 With this method a failure to set up a signaling link triggers the Maintenance Function to assess the IP trunk immediately Assuming the failure to set up the signaling link is the result of a network outage the Maintenance Function detects this and puts the signaling group out of service within one minute For example suppose there is an IP trunk between an S8700 system and an S8300 media gateway There is an outage in the IP network between the two systems and the 8700 discovers this after a measurement interval IP trunk bypass feature The S8700 puts the signaling group in bypass state and begins using the fallback TDM trunk The S8300 media gateway normally does not detect the outage until the next Maintenance Function cycle However if the 8300 attempts to place a call over the IP trunk and cannot establish a signaling link to the other end this triggers the Maintenance Function immediately which takes the signaling group out of service causing the fallback TDM trunk to be used So the S8300 detects the outage less than one minute after the first call attempt The scenario for severe congestion is different In the case of severe congestion the 8700 detects the congestion and puts the signaling group in bypass state the same as with a network outage It then sends a message to the S8300 indicating this condition This message is also sent in the network outage case but it doesn
73. d the DHCP requests shift to the data VLAN for the value of VLANTEST seconds If it is not answered it will shift back to the voice vlan for the same value of VLANTEST seconds DHCP requests will continually alternate between data and voice VLANs You no longer have to manually reset values to clear the information from memory Using VLANTEST 0 still works the same It will disable moving from the voice VLAN back to the data VLAN by keeping all requests on the voice VLAN Appendix A describes how to configure a simple network for dual VLAN operation Remember that in order for the CoS markings to have any effect the corresponding QoS configurations must be implemented on the necessary network devices Remember also that improperly enabling L2 and L3 prioritization may break processes that were working without it Read section 2 3 of this document for more information on CoS and QoS 4 3 Gatekeeper Lists and DHCP Option 176 An IP telephone can have a list of gatekeepers C LANs and S8300s or S8500 Main Servers to which it may send the initial RAS Gatekeeper Request GRQ message This list is obtained via the DHCP option 176 string which is covered briefly in section 4 1 and in detail in the LAN Administrator s Guide Within the DHCP option 176 string the comma separated IP addresses that follow the MCIPADD parameter constitute a gatekeeper list and this list provides redundancy at boot up Ifa given gatekeeper is unreachable for any
74. d to verify that their performances have not degraded to unacceptable levels Traffic Shaping on Frame Relay Links Experience to date supports Cisco s requirement to use traffic shaping on frame relay links 2 p 5 22 Simply stated VoIP traffic must be sent within the committed information rate CIR and not in the burst range This means that everything traversing a specific interface or sub interface must be sent within CIR because there is no mechanism to dictate that VoIP be sent within CIR while data is sent in the burst range on the same interface Under this constraint one solution for maximizing bandwidth is to make the CIR as large as possible and this is dictated by the end of the PVC that has the smaller access circuit Consult each router vendor s documentation to see if other methods are available KW Avaya IP Telephony Implementation Guide 26 3 Guidelines for Avaya Servers and Gateways This section gives guidelines for Avaya servers and gateways and covers most of the IP telephony related configurations Refer back to section for an overview of IP telephony components and Avaya architectures Avaya Communication Manager 1s the call processing software that runs on Avaya servers and it is configured via the Switch Administration Terminal SAT interface Although the server platforms themselves may be configured in various ways SAT is the universal interface for Communication Manager The Avaya Site Administrator
75. de the S8300 is a standalone call server In LSP mode it is a backup to the primary call KW Avaya IP Telephony Implementation Guide 28 server and must be activated An LSP does not accept station registrations or assume call processing responsibilities until 1t becomes active which occurs when a gateway registers to it The S8300 SAT interface may be accessed using Avaya SA ASA or by telnet ing to port 5023 telnet lt S8300 address gt 5023 or SSH ing to port 5022 This could also be done by telnet ing to the 8300 and typing sat from the Linux shell S8300 ICC permits SAT configuration changes and displays but S8300 LSP does not displays only because it receives its Avaya Communication Manager translations from the primary server The S8300 connects to the G700 G350 G250 via a backplane 100M Ethernet interface which is not configurable 3 3 G700 G350 G250 Media Gateways G700 P330 C360 L2 Switch The P330 L2 switch is the base platform for the G700 All other logical physical IP components MGP VoIP media modules LSP are connected to the P330 L2 switch The asynchronous port 9600 8 N 1 marked CONSOLE on the face of the G700 connects the user to the P330 CLI The IP expansion slot on the lower left corner of the chassis accepts the same X330 expansion modules used by the P330 switch The most common ones are probably the WAN router module and the 16 port Ethernet module The two Ethernet ports marked EXT1 and EXT2 are L2 switch port
76. diately Avaya P550 calls this fast start and Cisco calls it portfast If this feature 1s not available disabling STP on the port is an option that should be considered Do not disable STP on an entire switch or VLAN Enable Rapid Spanning Tree and configure host ports as edge ports As the name implies Rapid Spanning Tree Protocol RSTP is a faster and more advanced replacement for STP RSTP is preferred over STP when all network devices in a L2 domain support it Even if they don t there are ways to combine RSTP and STP depending on the network equipment though certainly not as clean as having RSTP throughout the L2 domain When running RSTP configure the host ports as edge ports which is equivalent to enabling fast start or portfast in a STP domain Disable Cisco features Cisco features that are not required by Avaya endpoints are auxiliaryvlan except for IP phones in a dual VLAN setting as described in appendices A and B channeling cdp inlinepower and any Cisco proprietary feature in general Explicitly disable these features on ports connected to Avaya devices as they are non standard mechanisms relevant only to Cisco devices and can sometimes interfere with Avaya devices The CatOS command set port host lt mod port gt automatically disables channeling and trunking and enables portfast Execute this command first and then manually disable cdp inlinepower and auxiliaryvlan For dual VLAN IP telephone implementations see Ap
77. dual VLAN implementation the VLANTEST parameter has great significance as illustrated below The following scenario with arbitrary voice VLAN ID details the steps a phone 1 8 and later would go through in a typical dual VLAN implementation It also illustrates the recommended content of the option 176 string Phone with no previously stored values boots up and obtains an address on the data VLAN The data VLAN option 176 string directs the phone to go to voice VLAN 25 MCIPADD adadrlI addr72 HT TPSRVR addr L2QVLAN 25 L2QAUD 6 L2QSIG 6 VLANTEST 600 Phone releases the data VLAN address and obtains an address on the voice VLAN The voice VLAN option 176 string is identical to the data VLAN string but without the LAQVLAN parameter because a phone already on the voice VLAN doesn t need to be directed to go there MCIPADD adadrl addr2 MCPORT 1719 TFTPSRVR addr L2QAUD 6 L2QSIG 6 VLANTEST 600 Phone is operational on the voice VLAN Reboot or power cycle occurs Phone immediately returns to voice VLAN 25 upon recovery and one of the following occurs Phone obtains an address and option 176 string on the voice VLAN and all is well Phone cannot obtain an address on the voice VLAN due to network or DHCP problems In this case the VLANTEST 600 parameter directs the phone to continue trying for 600sec finite range is 1 999 If the phone does not succeed in obtaining an address within 600sec it marks VLAN 25 as invalid and
78. e call server If configured locally the set qos commands are used to administer the settings There is no need and no parameter to set the VLAN ID because the MGP is already assigned to a VLAN via the set interface mgp command and all modules inherit that VLAN ID G700 in Octaplane Stack vs Standalone The G700 can be placed within an Avaya P330 Octaplane stack which provides 4Gig full duplex uplinks between the Avaya switches in the stack There are pros and cons to this The pros are that the entire stack can be managed as one unit via a single IP address there 1s abundant bandwidth between the switches in the stack and the entire stack can be uplinked to other network equipment without uplinking each individual switch in the stack The cons are that the initial configuration can be a little more complex and a problem associated with the stack can adversely affect the G700 Many factors can drive the decision to use or not use the Octaplane Device and uplink management are key factors If several G700 gateways are co located in the same rack it makes practical sense to use the Octaplane stacking feature This allows the P330 components of all the G700s to be managed via a single inband interface But more importantly it eliminates the need for each G700 to be uplinked to the next network device individually When determining whether or not a single G700 should be added to an existing Octaplane stack of P330 switches the relative importance
79. ected and often achieved but not always stable Suitable for user PC connections but not suitable for server connections or auto negotiate auto negotiate uplinks between network devices Suitable for a single VoIP call such as with a softphone or single IP telephone Not suitable for multiple VoIP calls such as through a MedPro MR320 board 100 half stable Devicel senses the speed and matches accordingly auto negotiate 100 half Devicel senses no duplex negotiation so it goes to half duplex auto negotiate 10 half 10 half stable Devicel senses the speed and matches accordingly Devicel senses no duplex negotiation so it goes to half duplex Devicel goes to 100 half resulting in a duplex mismatch auto negotiate 100 full undesirable Devicel senses the speed and matches accordingly Devicel senses no duplex negotiation so it goes to half duplex Avaya IP Telephony Implementation Guide 17 100 full 100 full 100 full stable Typical configuration for server connections and uplinks between network devices 10 half 10 half Stable at respective speed and duplex Some enterprises do this on 100 half 100 half user ports as a matter of policy for various reasons Table 1 Speed duplex matrix Layer L1 errors such as runts CRC errors FCS errors and alignment errors often accompany a duplex mismatch If these errors exist and continue to increment there is probably a duplex mismatch or cabling problem or some other physical laye
80. ecuted on a port by port basis sets the port speed sets the port duplex enables Spanning Tree fast start feature no to undo sets the native vlan default vlan when port is in access mode default is access mode where there is only one vlan on port puts port in trunk mode makes trunk 802 1Q instead of ISL specifies vlans permitted on trunk port default is all vlans sets the native vlan default vlan when port is in trunk mode Avaya IP Telephony Implementation Guide 69 Avaya P550 580 and P880 882 Switches show running config displays all configurations currently running on switch show startup config displays all configurations in NVRAM to be used at next boot up copy running config startup config must be executed to save running configuration to NVRAM not necessary on P330 switches except on router module set port auto negotiation lt mod port gt enables or disables speed duplex negotiation for given port s set port speed lt mod port gt sets the speed for given port s set port duplex lt mod port gt sets the duplex for given port s show port status displays settings and status for all ports or given port s show port counters displays high level TX and RX statistics for all ports or given port s show ethernet counters displays detailed statistics and errors for all ports or given port s clear port counters clears statistics and error counters on all ports or given port s set port fast start
81. ed primarily for VLAN trunking and the Priority field was not important The VID field is used as it always has been to indicate the VLAN to which the Ethernet frame belongs Rules for 802 1p Q Tagging There are two questions that determine when and how to tag 1 Is tagging required to place the frame on a specific VLAN VLAN tagging 2 Is tagging required to give the frame a priority level greater than 0 priority tagging Based on the answers to these questions tagging should be enabled following these two rules l i VLAN Ethernet switch port default scenario On a single VLAN port there is no need to tag to specify a VLAN because there is only one VLAN For priority tagging only the IEEE 802 1Q standard specifies the use of VID 0 VID 0 means that the frame belongs on the port s primary VLAN which IEEE calls the port VLAN and KW Avaya IP Telephony Implementation Guide 21 Cisco calls the native VLAN Some Ethernet switches do not properly interpret VID 0 in which case the port native VID may need to be used but this is not the standard method For single devices such as a call server or port board a simpler alternative is to not tag at all but configure the Ethernet switch port as a high priority port instead This treats all incoming traffic on that port as high priority traffic based on the configured level For multiple devices on the same VLAN such as an IP telephone with a PC attached the high
82. eeeessseeeseeeeesssaaees 78 FRE MCHOMCES aeae ian catatonia bits be sag aaanas E 80 KW Avaya IP Telephony Implementation Guide 6 1 Introduction to VoIP and Avaya Products This section provides a foundation to build upon for the rest of this document Voice over IP VoIP terminology and Avaya products and architectures are introduced here 1 1 Servers Gateways Stations and Trunks Defined Servers Most of the intelligence in a voice system is in the call server From servicing a simple call to making complex decisions associated with large contact centers the call server is the primary component of an IP telephony system Avaya Communication Manager is the call processing software that runs on Avaya server platforms The following are some common terms for a call server Some are generic and some are specified by a protocol but all are generally used throughout the industry Call Server generic term Call Controller generic term Gatekeeper H 323 term Media Gateway Controller H 248 term Gateways A gateway terminates and converts various media types such as analog TDM and IP A gateway is required so that for example an IP phone can communicate with an analog phone on the same telephony system as well as with an external caller across a TDM trunk The following are some common terms for a gateway and they are generally used throughout the industry Gateway generic and H 323 term Medi
83. egular and retry intervals Each keepalive mechanism has a regular interval as described above If a regular interval keepalive is not acknowledged more keepalives are sent at a faster retry interval If all the retry keepalives are unanswered the phone effectively unregisters and moves on to the next gatekeeper in its gatekeeper list obtained via DHCP and or the gatekeeper TTL As stated above the gatekeeper sends a TTL for the RAS keepalive mechanism The TTL is the greater of 60 seconds or a multiplier times the number of registered endpoints The multiplier for a CM server is approximately 1 4 seconds which means that anything above 42 registered endpoints would exceed the minimum 60 sec TTL The multiplier for the other servers described in this document is 1 second which means that more than 600 registered endpoints are required to exceed the minimum 60 sec TTL Independent of the mechanism RAS or TCP the keepalive flow follows this pattern regular interval regular interval retry int retry int retryint retryint retryint retry int Q KA KA D KA retry KA retry KA retry KA retry KA retry KA D ACK ACK 5 no ACK noACK noACK noACK noACK noACK 2 4 time to unregister _ ________ Q Figure 17 Keepalive pattern The discovering at the end of the flow means that the phone has effectively unregistered and is searching for another gatekeeper Effectively unregistered means that the phone has not sent an
84. en Avaya systems requires less C LAN resources than if each individual call has its own connection The greater the usage of the signaling group frequency of calls features utilized during calls number of simultaneous calls etc the greater the C LAN resource consumption As a very general rule based on anecdotal evidence of typical IP trunk usage and assuming calls share IP signaling connection substitute one signaling group for ten IP stations in the two preceding bullet items Another option is to dedicate C LAN boards and network regions specifically for IP trunks C LAN and MedPro MR320 Protocols and Ports Call signaling and media conversion between analog TDM and IP are key IP telephony functions The S8700 S8500 and DEFINITY servers use distributed C LAN boards to front end the call signaling and distributed MedPro MR320 boards to perform the media conversion The following table lists the protocols and ports used by both boards Section 3 5 heading ip network region gives instructions on how to configure the MedPro MR320 UDP port range See Appendix D for guidelines on configuring access lists UDP 1719 H 225 RAS IP station registration TCP 1720 H 225 Q 931 call signaling for IP C LAN stations and IP trunks TCP 2945 H 248 media gateway control signaling TCP 1039 H 248 encrypted MG control signaling MedPro UDP 2048 65535 configurable RTP encapsulated audio MR320 Table 6 C LAN and MedPro MR3
85. ence has shown that a 20 ms packet network is a good compromise between audio quality and bandwidth consumption Reducing to 10ms doubles the number of packets put onto the network but only 10ms of audio can be lost when a packet fails to reach its destination or arrives out of order Going beyond 20ms reduces the number of packets put onto the network but there is greater potential for poor audio quality when there is high packet loss Larger packets work better in low loss high jitter networks Smaller packets work better in high loss low jitter networks 20 ms packets are a good compromise 20 ms packet size recommended The Media Encryption portion of this form is an ordered list of preferred media encryption options For example an ordered list of AES AEA and none means that AES encryption is preferred first then AEA encryption if AES is not possible then no encryption 1f neither AES nor AEA is possible This list may contain one or more items Allow Direct IP Multimedia has to do with video over IP which is beyond the scope of this document For information on the remaining FAX Modem TDD TTY and Clear channel parameters see the product documentation Administration for Network Connectivity for Avaya Communication Manager 555 233 504 chapter 3 heading Administering FAX modem TTY and H 323 clear channel calls over IP trunks See also the document Avaya FoIP MoIP amp TTYoIP at www avaya com ip
86. end LRQ Wait 2sec for LCF 1sec as of Communication Manager 3 0 Send LRQ Wait 2sec for LCF 1sec as of Communication Manager 3 0 Go to next preferred trunk in route pattern 4sec total per call for Communication Manager pre 3 0 2sec total per call as of 3 0 The LRQ feature affects individual calls whereas the IP trunk bypass feature affects entire IP trunks The IP trunk bypass feature takes some time to detect a problem in the IP network and put the signaling group into bypass state When this happens with the appropriate route pattern in place it results in all calls being routed onto the next preferred trunk The LRQ feature speeds up per call re routes until IP trunk bypass is established so the two features can work in conjunction When LRQ is enabled the near end listen port must be 1719 This means that the far end gatekeeper must have its far end listen port set to 1719 If the far end gatekeeper is an Avaya call server and also has LRQ enabled near end listen port is 1719 then the near end gatekeeper must have its far end listen port set to 1719 Also when LRQ is enabled calls cannot share the IP signaling connection so this parameter must be set to n Each call establishes signaling across the IP trunk after a successful LRQ LCF exchange For information about IP trunking with the Cisco Call Manager see Avaya S8300 Media Server and Avaya S8700 Media Server Networked with Cisco Call Manager using H 323 Si
87. ent interface via a L3 router Like any other IP host the inband interface needs a default route if it is to route off of its VWLAN subnet The default route for the inband interface is displayed and configured using the commands show ip route and set ip route respectively If there is more than one router on the inband VLAN subnet the inband interface may have additional routes based on destination subnets or hosts These are displayed and configured using the same commands Finally the P330 L2 switch itself has various configuration parameters such as Spanning Tree VLANs trunking and speed duplex These are configured just like on the P330 switch see appendix E G700 Media Gateway Processor MGP The media gateway processor MGP is the media gateway portion of the G700 The MGP manages the various media modules inserted into the G700 These media modules include analog port modules for analog phones DCP port modules for DCP phones DS1 modules for TDM trunks and others The media module associated specifically with IP telephony is the VoIP module Each VoIP module is equivalent to a MedPro board and has 64 audio resources A single VoIP module is built in to the MGP and external VoIP modules can be added as necessary KW Avaya IP Telephony Implementation Guide 29 Like the P330 inband management interface the MGP should be thought of as a host on the P330 L2 switch The command session mgp from the P330 CLI puts the user into the MGP
88. ent times See the Avaya Communication Manager Network Region Configuration Guide at www avaya com for detailed instructions on configuring these parameters When connecting a gateway to another Ethernet switch the uplink between the two switches should be fixed at 1O0M full duplex see section 2 1 heading Speed Duplex Furthermore if 802 1p Q tagging information VLAND ID and or L2 priority is to be passed across the uplink both switches must have 802 1Q trunking enabled with matching VLANs on the connected ports For survivability reasons at remote offices the LSP media gateway and local IP phones should all be on the same voice VLAN subnet If the LSP media gateway and IP phones are on different subnets they depend on a router or routing process to function This is not desirable especially at a branch office where there is typically only one router The IP telephony system should be able to function even if that router or routing process fails In larger remote offices it may not be feasible to put all VoIP endpoints on a single subnet but the principle of minimizing dependencies still holds For remote offices where the WAN link terminates on the gateway whether on the X330WAN router or natively on the G350 G250 itself the DSCP values for audio and signaling must be 46 and 34 respectively The X330WAN router in voip queue mode and the G350 G250 gateways are optimized to use these values for QoS on the WAN link
89. er address The Status Summary web screen shows the status of the servers The S8700 SAT interface may be accessed using Avaya Site Administration ASA or by telnet ing to port 5023 telnet lt active server address gt 5023 A SAT session can also be established by telnet ing to the active server and typing sat from the Linux shell The standby server does not permit access to SAT Secure Shell SSH access is recommended for encrypted connections to an S87XX server pair ASA supports Secure Shell SSH access for system administration A SSH Client can also be used to access the SAT on port 5022 SAT access to the S8500 is similar to that of the S87xx server pair except that there is only one server As of CM 3 1 a S8500 main server or S8500 LSP supports Processor Ethernet PE The S8500 PE provides similar functions as a CLAN TN799DP Circuit Pack for H 323 IP endpoints H 248 gateways and subset of adjuncts An S8500 PE interface uses one of the native NICs on the server and allows for direct connections to H 248 Media Gateways without the need for port networks CLAN IPSI There are however configuration limitations which are defined in the Overview for AVAYA Communication Manager Document ID 03 300468 available on the support avaya com website S87xx S8500 Speed Duplex Speed and duplex for the various S87xx S8500 Ethernet interfaces are configured using the Configure Server Configure Interfaces web admin screen It is critic
90. ernet switch The built in 10 100 Ethernet switch permits speed and duplex configuration if necessary This switch is set to auto negotiate speed and duplex by default The closest Ethernet switch to which the IP Telephone is attached should be set to auto negotiate as well Locking down the closest switch to full duplex without also locking down the duplex of the phone will lead to packet loss and thus result in problems with voice quality Follow the guidelines in section 2 1 heading Speed Duplex when configuring the speed and duplex on these phones and the Ethernet switch ports to which they are connected Current models also have an updated look and a larger screen that facilitates additional features and functionality Feature related implementation and the additional features and functionality of the current models are covered in IP telephone specific documentation found at support avaya com When the IP phone and PC are both transmitting the phone s traffic is given strict priority out the uplink port to the enterprise Ethernet switch This 1s not an issue for the PC because under normal conditions the IP phone transmits less than 100kbps of audio traffic Prioritization of traffic downstream from the enterprise Ethernet switch to the phone s switch port must be handled by the enterprise Ethernet switch The built in Ethernet switch strips the 802 1Q tag from the IP telephone toward the PC That is tagged traffic from the phone
91. et when the packets were transmitted The PADDING flag indicates the presence of padding octets at the end of the RTP payload which cannot be true for G 711 Why this occurred is unknown but it does not really matter because there is no point in using the G 711 codec if bandwidth is scarce Configuration To configure RTP header compression on a Cisco router 1 Specify the number of RTP connections that can be compressed cache allocation In interface configuration mode the command is ip rtp compression connections lt number gt The default is 32 and each call requires two connections The configurable range is 3 to 256 for PPP and HDLC using IOS v11 3 and later and 3 to 1000 for PPP and HDLC using IOS v12 0 7 T and later For Frame Relay the value is fixed at 256 2 The command to turn on compression is ip rtp header compression in interface configuration mode It must be implemented at both ends of the WAN link For this experiment when the command was entered into the router ip tcp header compression was also installed automatically When either command was removed the other was automatically removed Consult Cisco s documentation for more specific configurations on other types of WAN links ie Frame Relay and ATM 2 p 5 14 5 18 5 26 5 33 3 Configuration for the X330WAN router is very similar to Cisco and well documented in the X330WAN User Guides KW Avaya IP Telephony Implementation Guide 66 Appendix D Ac
92. etary mechanisms and are not subject to constraint by a standards body or by Avaya 802 1Q trunking is well tested successfully deployed and defined by a standards body but the configuration is not as clean and trunking on user ports has other network implications For IOS based Catalyst switches voice vlan is roughly equivalent to auxiliaryvlan On older IOS platforms ie 2900XL 3500XL there appears to be no configuration or functionality benefit to using voice vlan as explicit trunking is still required when voice vlan is enabled on these older platforms On newer IOS platforms ie 3550 3560 however voice vlan can be enabled without explicit 802 1Q trunking so there are benefits to using voice vlan on these newer platforms Note that Avaya IP phones do not interoperate with CDP Therefore although auxiliaryvlan and voice vlan can be used the mechanism of discovering these VLANs via CDP is not supported The Avaya IP phone can learn the auxiliaryvlan voice vlan designation via DHCP option 176 as explained below and in Appendix A How it Works KW Avaya IP Telephony Implementation Guide 62 The remainder of this document focuses on auxiliaryvlan CatOS but voice vlan IOS operates on the same principles as auxiliaryvlan At the heart of Cisco s auto discovery feature are Cisco proprietary mechanisms The first proprietary mechanism is CDP Cisco Discovery Protocol This is a layer 2 protocol which means that it works
93. ffic between S87xx S8500 and IPSI board Permit IPSI board IP any Permit IP any TCP 80 TCP any TCP 443 TCP any UDP 53 dns UDP any KW Avaya IP Telephony Implementation Guide 68 Appendix E Common IP Commands Cisco CatOS Switches set port speed lt mod port gt set port duplex lt mod port gt show port show port lt mod port gt clear counters set port host clear port host set spantree portfast lt mod port gt show spantree lt mod port gt set vlan lt vlan id gt lt mod port gt set port auxiliaryvlan lt mod port gt lt vid gt set port auxiliaryvlan lt mod port gt none show port auxiliaryvlan set trunk all off set trunk lt mod port gt clear trunk lt mod port gt clear trunk lt mod port gt lt vid s gt show trunk OR show port trunk show vlan sets the speed for given port s sets the duplex for given port s displays settings and status for all ports displays settings statistics and errors for given port clears statistics and error counters on all ports or given port s disables channeling trunking enables portfast on all or given port s opposite of set port host enables or disables Spanning Tree fast start feature on given port s displays Spanning Tree and portfast info for all ports or given port s sets the native vlan default vlan for given port s sets the auxiliary vlan for given port s removes auxiliary vlan from given port s d
94. g to be enabled to accept VID 0 while others do not The following table shows the results of some testing performed by Avaya Labs on Cisco switches Catalyst 6509 w Accepted VID 0 for the native VLAN when 802 1Q trunking was enabled CatOS 6 1 2 on the port Would not accept VID 0 for the native VLAN Opened a case with Cisco Catalyst 4000 w TAC and TAC engineer said it was a hardware problem in the 4000 Bug CatOS 6 3 3 ID is CSCdr06231 Workaround is to enable 802 1Q trunking and tag with native VID instead of 0 IOS 12 0 5 WC2 on the port documentation or call TAC Table 4 Sample VID 0 behaviors for Cisco switches See Appendix A for more information on VLANs and tagging KW Avaya IP Telephony Implementation Guide 22 DSCP IP Header lt 32 bits gt octet octet octet octet Header l Version Length Type of Service Total Length Identifier Fragment Offset Time to Live Protocol Header Checksum Source Address Destination Address Options and Padding Figure 14 IP header The figure above shows the IP header with its 8 bit Type of Service ToS field The ToS field contains three IP Precedence bits and four Type of Service bits as follows Routine Priority Bits 0 2 Immediate Flash IP Precedence Flash Override CRITIC ECP Internetwork Control Network Control Bit 3 Delay Throughput Reliability Monetary Cost Reserved Figure 15 Original scheme fo
95. ge PNC and ATM PNC are still present to connect the port networks S8500 Media Server pare 4 Avaya 8500 Media Sarit The Avaya S8500 Media Server is the simplex equivalent of the S87xx server pair The S8500 gives the same call processing capability without the redundancy and added reliability of duplicated servers The S8500 can be substituted in place of the S87xx servers in any IP Connect configuration shown below KW Avaya IP Telephony Implementation Guide 10 IP Connect Medium Large Enterprise 58700 Ts Enterprise IP Network CCMS over TCP IP Analog DCP Figure 5 IP Connect 38300 G700 G350 G250 The Avaya G700 G350 and G250 not shown Media Gateways are based on the H 248 protocol One primary difference between these gateways is capacity Refer to current product offerings for exact specifications All gateways have built in Ethernet switches The G700 supports IP routing and IP WAN connectivity with an expansion module and the G350 and G250 support them natively The G700 is built on the Avaya P330 Stackable Switching System with similar CLI The G350 and G250 are built on a new IP platform also with similar CLI All gateways use compact media modules instead of traditional port boards The VoIP media module serves the same function as the MedPro board There are other media modules equivalent to traditional port boards analog DCP DS1
96. gnaling and IP Trunk Groups at www avaya com KW Avaya IP Telephony Implementation Guide 42 media gateway Options are add change display and list This form is used to administer a G700 G350 G250 media gateway Number is simply a numeric index Type is the media gateway model ie G700 G350 G250 G250 BRI Name is a text descriptor Serial No is the gateway s serial number which is displayed by typing show system at the MGP CLI A gateway must be administered on the call server before it can register to that server and the serial number is what uniquely identifies a valid gateway Network Region is used for IGAR purposes similar to assigning port networks to a network region on the cabinet form But unlike the cabinet form the network region designation on the media gateway form also assigns the gateway VoIP resources to a particular Communication Manager network region This is equivalent to assigning a MedPro MR320 board to a network region on the ip interfaces form Recovery Rule determines automatic recovery back to the primary server while the media gateway is registered to an LSP The default is no automatic recovery none or a number can be placed here to apply a recovery rule per the system parameters mg recovery rule form as explained in the following section Encrypt Link refers to the H 248 signaling link between the gateway and the call server Location serves the same function as the identical field on the cabinet
97. granularity required to gauge network performance Because pings are used to determine network performance the IP network should ideally give the pings ICMP Echoes and Echo Replies between MedPros MR320s and C LANs the same priority as audio traffic To facilitate this it is important to know that the call server can select any MedPro MR320 in the near end system s network region to originate the pings Depending on the network it may be feasible to activate this feature without deploying any network policies for the pings especially if the primary concern 1s to compensate for network outages and not necessarily for poor performance Q2 Besides the IP trunk bypass feature what other mechanisms are in place to detect an outage or severe congestion in the IP network and how long does it take to detect it KW Avaya IP Telephony Implementation Guide 75 See section 3 5 heading trunk group and signaling group for details on the LRQ feature that applies to individual calls placed over an IP trunk For the IP trunk as a whole the best method is the IP trunk bypass feature In addition there is also a Maintenance Function This function assesses the IP trunk every 15 minutes in a G3r or Linux platform and every hour in a G3i platform Without going into detail the Maintenance Function determines whether the signaling group is in service or out of service It can detect a network outage but it does not assess network performance A thir
98. h periods of poor audio quality then there is probably a speed duplex problem between the board and the Ethernet switch The maximum MR320 throughput is C LAN and MedPro MR320 802 1p Q and DSCP See section 3 5 headings 1p interface and 1p network region L2 and L3 prioritization on the C LAN requires the TN799DP board with firmware v5 or later MR320 Capabilities and MR320 Bearer Duplication The TN2602AP IP Media Resource 320 provides either 80 or 320 encrypted or unencrypted channels of 2 way audio RTP streams or conversations Channel capacity is dependent on software licensing The MR320 supports G 711 G 729A B and G 726A codecs See Table 6 for a comparison of Medpro and MR320 capabilities A single port network can have up to two TN2602AP circuit packs only As result the port network can have either two duplicated TN2602AP circuit packs or two load balancing TN2602AP circuit packs but not both a duplicated pair and a load balancing pair However in a Communication Manager configuration some port networks can have a duplicated pair of TN2602AP circuit packs and other port networks can have a load balancing pair of TN2602AP circuit packs and some port networks can also have single or no TN2602AP circuit packs The TN2602AP IPMedia Resource 320 can provide duplicated bearer for IP Connected Port Networks This enables customers to administer IP PNC with critical bearer reliability A port network supports a maximum of two
99. h to isolate to a port range The GET and PUT requests from the client go to the server s UDP port 69 but all other messages go between random ports Permit UDP 53 dns Permit UDP any Permit UDP 67 bootps Permi UDP 68 bootpc Permit Permit Permit TCP 411 Permit TCP any Permit TCP 80 81 Permit TCP any Permit UDP 161 snmp Permit Any IP telephone hardphone UDP 161 snmp SNMP management station s UDP any er KW Avaya IP Telephony Implementation Guide 67 Avaya devices ping other devices for various reasons For example C LANs ping endpoints for management purposes MedPros MR320s ping C LANs to gauge network performance across an IP trunk IP telephones ping TFTP servers for verification purposes Any Avaya device ICMPEcho Any S de o Permit Any ICMP Echo Any Avaya device Reply The following table contains access list guidelines for Avaya media servers and media gateways Most connections take place over the S8xxx server s enterprise interface which could be a separate interface or combined with a control network interface The enterprise interface is Eth4 on S87xx Multi Connect Typically ethO on S87xx IP Connect but could also be configured as eth4 in some cases Typically ethO on S8500 but could also vary by configuration Action From TCP UDP port To TCP UDP port or Protocol or Protocol This allows the Communication Manager 1 x primary server to synchronize translations wi
100. he branch there should also be a dedicated DHCP server because IP telephones require both services For cost and administrative reasons however many will choose not to install a DHCP server at all branch locations In such cases it is very important that the IP telephones not be rebooted during a WAN link failure because they would not be able to obtain IP addresses The manual configuration option 1s available but it is not always a viable option for various reasons Two Methods of Receiving the Gatekeeper List In addition to receiving the gatekeeper list via DHCP option 176 as described above a gatekeeper list is also received via the RCF message during registration In other words when an IP telephone registers with the call server the call server sends a gatekeeper list in the RCF message The H 323 standard calls this the Alternate Gatekeeper List This means that a phone really only needs one gatekeeper address at boot up because the phone receives the gatekeeper list when it registers This feature is useful for phones that are manually administered as the manual method only permits the entry of one gatekeeper address However it is still preferable to administer a gatekeeper list in DHCP option 176 for redundancy during boot up Here are some key points regarding the option 176 gatekeeper list and the RCF Alternate Gatekeeper List IP telephone versions prior to 2 0 use both lists simultaneously GK addresses received from eithe
101. he most prevalent parameters and values are as follows MCIPADD Address es of gatekeeper s at least one required MCPORT The UDP port used for registration 1719 default TFTPSRVR Address es of TLS HTTP S TFTP server s at least one TLSSRVR required HTTPSRVR L2QVLAN 802 1Q VLAN ID 0 default L2QAUD L2 audio priority value L2QSIG L2 signaling priority value VLANTEST The number of seconds a phone will attempt to return to the previously known voice VLAN Table 8 DHCP option 176 parameters and values The typical option 176 string for a single VLAN environment looks like this MCIPADD adadr1 addr2 addr3 MCPORT 1719 HTTPSRVR addr At least one gatekeeper C LAN or S8300 or S8500 Main Server address must be present after MCIPADD to point the phones to a call server MCPORT specifies which UDP port to use for RAS registration IP telephone firmware 1 6 1 and later already have 1719 as the default port but it is prudent to include it A TFTP server address is necessary so that phones know where to go to download the necessary script files and binary codes see Boot up Sequence heading below LZ2QVLAN and VLANTEST would be included if 802 1Q tagging were required such as in a dual VLAN environment see section 4 2 Other parameters may be added such as LZQAUD and L2QSIG which are used to specify the L2 priority values for audio and signaling If these values are not specified in option 176 the defaul
102. hin the LAN will update their old mapping in ARP cache with this new mapping It is also possible to invoke an interchange manually via a software command Duplicated TN2602AP circuit packs must be in the same subnet In addition the Ethernet switch or switches that the circuit packs connect to must also be in the same subnet This shared subnet allows the Ethernet switches to use signals from the TN2602AP firmware to identify the MAC address of the active circuit pack This identification process provides a consistent virtual interface for calls The Communication Manager license file must have entries for each circuit pack with the entries having identical voice channels enabled In addition both circuit packs must have the latest firmware that supports bearer duplication Capability TN2302AP HW Version 11 or later TN2602AP Codecs e G 711 64 maximum unencrypted e G 711 80 or 320 channels by license channels 48 maximum encrypted unencrypted or encrypted channels e G 729A G 729AB 80 or 320 e G 729B and G 723 1 32 maximum channels by license unencrypted or unencrypted 24 maximum encrypted encrypted channels e G 726A 80 or 320 channels by license unencrypted or encrypted Fax Relay proprietary 16 unencrpyted 12 encrypted 80 or 320 by license unencrypted or encrypted Extreme Measures for MedPro and Other IP Boards on Cisco Switches This information is intentionally placed here and not in section 2 1 bec
103. igurations there may not be a G 711 path and in such cases setting this parameter to y forces the use of G 711 for music transport IP DTMF Transmission Mode The intra system parameter determines how DTMF tones are passed within a system between media gateways and P connected port networks with no traditional PNC Center Stage or ATM The inter system parameter configured on the signaling group form determines how DTMF tones are passed between systems across IP trunks Note that both ends of the IP trunk must be configured the same The primary issue driving these parameters is the fact that DTMF tones are not accurately reproduced using compressed codecs This is particularly an issue for systems that rely on DTMF tones for functionality The options operate as follows in band If the configured codec is G 711 or G 729 the tones are passed in band Otherwise the tones are passed out of band via call signaling G 711 accurately passes DTMF tones while G 729 can pass the tones but is susceptible to error This option is obsolete on CM 3 1 for intra system DTMF digits in band g711 If the configured codec is G 711 the tones are passed in band Otherwise the tones are passed out of band via call signaling This option removes the uncertainty of G 729 This option is obsolete on CM 3 1 for intra system DTMF digits out of band The digits represented by the tones are always passed out of band If H 245 messages are exchanged
104. ion L2 Ethernet PPP frame relay ATM Figure 10 VoIP protocol stacks H 323 is the prevalent VoIP protocol suite It is used for signaling from gatekeeper to terminals stations and gatekeeper to gatekeeper trunks H 225 is the endpoint registration RAS and call signaling Q 931 component of H 323 H 225 call signaling messages are transported via TCP with port 1720 on the gatekeeper side H 225 registration messages commonly referred to simply as RAS message are sent via UDP with port 1719 on the gatekeeper side H 245 is the multimedia control component of H 323 Audio is digitally encoded prior to transmission and decoded after transmission using a coder decoder codec G 711 is the fundamental codec based on traditional pulse code modulation PCM and it is generally recommended for LAN transport G 729 is a compressed codec and it is generally recommended for transport over limited bandwidth WAN links Encoded audio is encapsulated in RTP real time protocol then UDP then IP RTP has fields such as Sequence Number and Timestamp that are designed for the transport and management of real time applications On Avaya solutions the UDP ports used to transport RTP streams are configured on the call server Most protocol analyzers can identify RTP packets making it easy to verify that audio streams are being sent between endpoints H 248 is a protocol for media gateway control It is transported vi
105. irecting an active call to a different MedPro MR320 or VoIP module in this type of failure Q9 How is call processing affected in general by an IP trunk outage If the IP trunk outage is the result of a C LAN S8300 failure direct IP IP calls remain up until one of the IP phone goes on hook If the IP trunk outage is the result of a MedPro MR320 VolP failure existing calls are affected as previously described If the IP trunk outage is the result of the IP network going down the audio is lost on active calls and new calls are routed over the fallback TDM trunk if one 1s administered KW Avaya IP Telephony Implementation Guide 77 Appendix H IPSI Signaling Bandwidth Requirements VoIP deployments require the provisioning of priority service for VoIP bearer traffic RTP and a guaranteed bandwidth service for Call Signaling General VoIP requirements are discussed in the AVAYA IP Voice Quality Network Requirements document available on the www support avaya com website The following list summarizes key QoS requirements for voice traffic Voice bearer traffic should be marked with DSCP 46 EF Voice bearer and Voice Signaling Packet loss should not be greater than 3 One way delay should be no more than 150 milliseconds Average one way jitter should be less than 30 milliseconds H 323 Signaling Traffic should be marked DSCP 26 AF31 IPSI Call signaling traffic should be given a guaranteed bandwidth on WAN links IPSI Signaling Packet lo
106. is concerned only with the IP specific configuration parameters On the trunk group form the Group Type should be isdn the Carrier Medium should be IP and each member s Port designation beginning on page 3 of the form should also be IP Once the members are used for active calls the call server automatically changes the port designations to T which are internal port numbers The number of members determines the number of simultaneous calls The signaling group parameters are as follows Group Type h 323 Remote Office n in most cases y if the far end is a G150 R300 or MultiVOIP gateway Trunk Group for Channel Selection Specify the trunk group configured as described above Near end Node Name The node name of the local gatekeeper C LAN or S8300 terminating the H 323 signaling link as defined in the local call server s node names ip and ip interface forms Near end Listen Port 1720 by default This is the default TCP port used by the gatekeeper for H 225 call signaling Far end Node Name The node name of the far end gatekeeper terminating the H 323 signaling link as defined in the local call server s node names ip form Far end Listen Port 1720 by default if far end gatekeeper is an Avaya server or Cisco Call Manager May vary from device to device if configured to listen on a different TCP port Far end Network Region The numeric identifier of the locally defined network region with which
107. isplays auxiliary vlan information disables trunking on all ports sets trunking mode for given port s puts given port in auto trunk mode with negotiating encapsulation removes specified vlans from given trunk port s all vlans are permitted on trunk by default displays trunking information for all ports or given port displays vlan configuration information Cisco IOS Switches Global commands show running config show startup config copy running config startup config show interfaces status show interfaces fast gig lt mod port gt clear counters fast gig lt mod port gt show controllers ethernet controller clear controllers ethernet controllers show vlan Interface commands speed duplex spanning tree portfast switchport access vlan lt vid gt switchport mode trunk switchport trunk encapsulation dotlq switchport trunk allowed vlan switchport trunk native vlan KW displays all configurations currently running on switch displays all configurations in NVRAM to be used at next boot up must be executed to save running configuration to NVRAM not necessary on CatOS switches except on router module displays settings and status for all ports displays port s status statistics and errors at the interface level clears show interfaces counters displays port s statistics and errors at the controller level clears show controllers counters displays vlan configuration information These commands are ex
108. ist may not be comprehensive as only key events are listed The packets described here can be captured using a protocol analyzer and one with H 323 capability is required to properly decode the H 225 RAS messages On 4606 12 24 models the analyzer can be attached to the phone s user port But because the 4620 and 4610 have a built in switch instead of a hub the analyzer must be attached to a mirrored Ethernet switch port or to a tap or hub in line between the phone and the Ethernet switch Initial startup At power up or manual reset the phone goes through a short initial startup procedure The display shows Restarting if the phone was intentionally restarted w Hold RESET and then Loading and Starting DHCP The phone queries the DHCP server for an IP address and other needed information The following packets are exchanged DHCP Discover from phone to broadcast DHCP Offer from server to broadcast or relay agent to phone DHCP Request from phone to broadcast and DHCP ACK from server to broadcast or relay agent to phone Note that this step is bypassed if the phone is manually configured with all the necessary information Request file 46xxupgrade scr and others from TLS HTTP TFTP server This is a text script file that tells the phone which boot code and application code are needed If the phone does not have the current codes it requests them from the file server A brand new phone makes all three requests as
109. lephony Implementation Guide 49 TCP keepalive The IP telephone sends TCP keepalive messages to the gatekeeper at a regular interval determined by the phone or as administered on the ip network region form The keepalive is an empty TCP datagram with a sequence number that is 1 to 5 less than the sequence number of the previous real TCP message or ACK sent by the phone The gatekeeper acknowledges each keepalive from the phone with a similar empty TCP datagram This exchange takes place over the call signaling socket which has TCP port 1720 on the gatekeeper side The CLAN sends TCP keepalives similar to the phone s TCP keepalives However because a CLAN must keep track of potentially hundreds of phones the CLAN s keepalive intervals are much longer than the phone s keepalive intervals A CLAN sends regular keepalives to every phone once every 10min These keepalives are not synchronized so they don t all go out to every phone at the same time If one of the 10 min keepalives is missed the CLAN sends five more retry keepalives 1min apart So a link bounce detection time for a CLAN is 5 15min Ifa phone becomes unreachable and does not re register for an extended period of time it takes the CLAN 5 15min to discover that the phone is no longer reachable This means it takes CM 5 15min to internally unregister that phone The detection time is much faster less than a minute if CM tries to deliver a call to that phone and fails R
110. lt mod port gt enables or disables Spanning Tree fast start feature on given port s show port lt mod port gt displays Spanning Tree and fast start info for all ports or given port s set port vlan lt mod port gt lt vid gt sets the port vlan default vlan for given port s set port trunking format lt mod port gt sets trunking mode for given port s set port vlan binding method lt md pt gt sets the vlan binding method for given port s show port lt mod port gt displays trunking and vlan binding info for all ports or given port s show vlan displays vlan configuration information Avaya P330 C360 Switches set port negotiation lt mod port gt enables or disables speed duplex negotiation for given port s set port speed lt mod port gt sets the speed for given port s set port duplex lt mod port gt sets the duplex for given port s show port lt mod port gt displays settings and status for all ports or given ports s show rmon statistics lt mod port gt displays statistics and errors for given port s must reset switch to clear these counters set port spantree lt mod port gt enables or disables Spanning Tree on given port no fast start on P330 show spantree lt mod port gt displays Spanning Tree information for all ports or given port set port vlan lt vid gt lt mod port gt sets the port vlan default vlan for given port s set trunk lt mod port gt sets trunking mode for given port s
111. ly discard packets in this queue based on DSCP lower values get discarded first interface Serial0 description T1 ip address 172 16 0 1 service policy output voipQoS apply the voipQoS policy outbound on this interface KW Avaya IP Telephony Implementation Guide 71 This is an example of an ideal scenario where audio and H 323 and IPSI signaling are put in separate queues and receive different treatment The X330WAN router and the G350 G250 integrated routers are optimized to use separate queues for audio and signaling When terminating a WAN link to these devices audio must be marked with DSCP 46 and signaling with DSCP 34 and IPSI signaling with 36 Example 2 Common large enterprise implementation It is somewhat common to put audio and signaling in the same queue in which case both audio and signaling would be marked with the same DSCP class map match any VoIP create a class map called VoIP match ip dscp 46 any packet with DSCP 46 is in this class class map match any IPSI create a class map called IPSI match ip dscp 36 any packet with DSCP 36 af42 is in this class policy map voipQoS create a policy map called voipQoS class IPSI guarantee bandwidth to IPSI traffic bandwidth 128 prioritize packets in the VoIP class and dedicate 816k class VoIP of this WAN link priority 816 put everything else in the default class and transmit it out the default class class default queue in a weighted fair queue fashion fair queue if the default que
112. ly v1 and v10 on this port port is an 802 1Q trunk port though not explicitly configured Explicit trunking is an option set trunk 1 5 nonegotiate dotlq Plain 802 1Q trunk port with no Cisco negotiation features clear trunk 1 5 2 9 11 1005 Unnecessary VLANs removed and 10 remain If the P330 C360 switch were a Cisco IOS switch instead All ports have port native VLAN 1 by default interface FastEthernet0 1 KW Avaya IP Telephony Implementation Guide 60 switchport trunk encapsulation dotlq switchport trunk native vlan 1005 switchport trunk allowed vlan 1 10 1005 switchport mode trunk spanning tree portfast interface FastEthernet0 2 switchport access vlan 10 spanning tree portfast switchport priority default 6 interface FastEthernet0 3 switchport access vlan 10 spanning tree portfast switchport priority default 6 interface FastEthernet0 S switchport trunk encapsulation dotlq switchport trunk native vlan 1 switchport trunk allowed vlan 1 10 switchport mode trunk spanning tree portfast interface FastEthernet0 S switchport mode access switchport access vlan 1 switchport voice vlan 10 IP phone configuration Port connected to Cisco router is an 802 1Q trunk port Cisco switches do not tag the native VLAN but the router expects a tag on v1 so the native VLAN is changed to some unused VLAN VLANs 1 10 and 1005 allowed on trunk Port is in trunk mode Spanning Tree fast start feature Port native VLAN
113. maining on the port native VLAN See the instructions in section 2 3 under the heading Rules for 802 1p Q Tagging The phone must be configured to apply the appropriate L2 and or L3 priority values The Hold ADDR menu is used to manually enable or disable 802 1Q tagging and to set the VLAN ID The other parameters are configured via the Hold QOS menu The manual method is covered below and an automated method is covered in the next paragraph 802 1Q On off for 802 1Q tagging Turn this on if L2 priority tagging is desired off otherwise VLAN ID Should be zero 0 for this scenario per the instructions in section 2 3 heading Rules for 802 1p Q tagging The VID has no effect when 802 1Q tagging is disabled VLANTEST Not relevant when VID is zero Applies in a dual VLAN environment when VID is a non zero value as explained in later sections L2 audio Layer 2 CoS tag for Ethernet frames containing audio packets The phone either receives this from DHCP most common or from the call server rare per the ip network region form This value could also be set manually on a per phone basis L2 signaling Layer 2 CoS tag for Ethernet frames containing signaling packets The phone either receives this from DHCP most common or from the call server rare per the ip network region form This value could also be set manually on a per phone basis L3 audio Layer 3 DSCP for audio IP packets The phone automa
114. ment regarding Cisco products and features are best effort attempts at summarizing and testing the information and advertised features that are openly available at www cisco com Although all reasonable efforts have been made to provide accurate information regarding Cisco products and features Avaya makes no claim of complete accuracy and shall not be liable for adverse outcomes resulting from discrepancies It 1s the user s responsibility to verify and test all information in this document related to Cisco products and the user accepts full responsibility for all resulting outcomes 2005 Avaya Inc All Rights Reserved Avaya and the Avaya Logo are trademarks of Avaya Inc or Avaya ECS Ltd a wholly owned subsidiary of Avaya Inc and may be registered in the US and other jurisdictions All trademarks identified by and are registered trademarks or trademarks respectively of Avaya Inc All other registered trademarks or trademarks are property of their respective owners KW Avaya IP Telephony Implementation Guide 2 Foreword Several benefits are motivating companies to transmit voice communications over packet networks originally designed for data Cost saving is one factor By eliminating a separate circuit switched voice network businesses avoid the expenses of buying maintaining and administering two networks They may also reduce toll charges by sending long distance voice traffic over the enterprise network rather than the
115. n Avaya P330 v3 2 8 and later Cisco CatOS and some Cisco IOS VLAN Defined With simple Eth switches the entire switch is one L2 broadcast domain that typically contains one IP subnet L3 broadcast domain Think of a single VLAN on a VLAN capable Eth switch as being equivalent to a simple Eth switch A VLAN is a logical L2 broadcast domain that typically contains one IP subnet Therefore multiple VLANs are logically separated subnets analogous to multiple switches being physically separated subnets A L3 routing process is required to route between VLANs just as one 1s required to route between switches This routing process can take place on a connected router or a router module within a L2 L3 Eth switch If there is no routing process associated with a VLAN devices on that VLAN can only communicate with other devices on the same VLAN For a tutorial and more information on VLANs see LANs and VLANs A Simplified Tutorial at www avaya com The Port or Native VLAN Port VLAN and native VLAN are synonymous terms The IEEE standard and most Avaya switches use the term port VLAN 6 p 11 but Cisco switches use the term native VLAN Issue the command show trunk on Avaya P330 C360 and Cisco CatOS switches to see which term is used 1n the display output Every port has a port native VLAN Unless otherwise configured it is VLAN 1 by default It can be configured on a per port basis with the following commands Avaya P330 an
116. nd its CDP packet immediately instead of waiting for the transmit period 60 seconds by default to recycle 1 p 2 23 Avaya IP Phones on Cisco Auxiliaryvlan The auxiliaryvlan is a modified method of implementing 802 1Q trunking and it may be nothing more than this Although testing to date has been positive Avaya does not know what other mechanisms are or will be incorporated with this feature or if they could have any adverse effects on Avaya IP phones Assuming that an auxiliaryvlan enabled port 1s truly a standard 802 1Q trunk port the following steps allow Avaya IP phones to work on Cisco s auxiliaryvlan 1 Verify that auxiliaryvlan is enabled a For example the command set port auxiliaryvlan 2 4 8 500 would make ports 2 4 through 2 8 auxiliaryvlan capable with auxiliaryvlan ID 500 b The command set port auxiliaryvlan 2 4 8 w o the 500 would make ports 2 4 through 2 8 auxiliaryvlan capable with the default auxiliaryvlan ID 200 c The command show port auxiliaryvlan reveals the ports that have been made auxiliaryvlan capable and their respective auxiliaryvlan ID s The command show port reveals each port s port native VID 2 Bring up the phones on the auxiliaryvlan using the same procedures that would be used on a regular trunk port a Verify that a L3 router interface exists for both the port native VLAN and the auxiliaryvlan with an associated subnet and gateway IP address Both interfaces must be configured to forward
117. nnot be mapped to any IP Precedence value alone Another hurdle is if the legacy code implemented IP Precedence with only one ToS bit permitted to be set high In this case a DSCP of 46 still would not work because it would require two ToS bits to be set high When these mismatches occur the legacy device may reject the DSCP marked IP packet or exhibit some other abnormal behavior Most newer devices support both DSCP and the original ToS scheme QoS on an Ethernet Switch On Avaya and Cisco Catalyst switches VoIP traffic can be assigned to higher priority queues Queuing is very device dependent but in general an Ethernet switch has a fixed number of queues that may be configurable but are typically defaulted to optimized settings for most implementations The number of queues and the technical sophistication of the queuing vary among switches but in general the more advanced the switch the more granular the queuing to service the eight L2 priority levels An Ethernet switch can classify traffic based on the 802 1p Q priority tag and assign each class of traffic to a specific queue but only if this is a default feature or it is explicitly configured On many switches a specific port can be designated as a high priority port causing all incoming traffic on that port to be assigned to a high priority queue This frees the endpoint from having to tag its traffic with L2 priority QoS on a Router It is generally more complicated
118. nterpret VID 0 as the port native VID per the IEEE 802 1Q standard 6 p 69 If the Ethernet switch does not understand VID 0 the phone may need to tag with the port native VID although this is not the standard method Remember that in order for the CoS markings to have any effect the corresponding QoS configurations must be implemented on the necessary network devices Remember also that improperly enabling L2 and L3 prioritization may break processes that were working without it Read section 2 3 of this document for more information on CoS and QoS IP Phone and Attached PC on Different VLANs The third and most common scenario for attaching a PC to the phone the first two were covered in the previous heading is to have the phone and PC on separate VLANs This requires a dual VLAN port on the Ethernet switch as described in section 2 3 heading Rules for 802 1p Q Tagging One of the VLAWNs is the port native VLAN the data VLAN and clear Ethernet frames ones with no 802 1Q tag from the PC are forwarded on this VLAN The other VLAN is the voice VLAN and the IP phone must tag its traffic with the proper VLAN ID to have it forwarded on this VLAN The Hold ADDR and Hold QOS menu options are the same as described in the previous heading except that the VID must not be zero The preferred method of using DHCP option 176 section 4 1 heading DHCP Option 176 is also the same except that LAQVLAN has a non zero value Finally in a
119. o exceed the recommended maximum broadcast rate 500 s and do not exceed the absolute maximum broadcast rate 1000 s In summary the general guidelines are All switched network no hubs Separate voice VLANs No more than 500 hosts 23 subnet mask on voice VLAN with only VoIP endpoints No more than 250 hosts 24 subnet mask each on voice and data VLANs if IP phones have PCs attached to them 100M uplink between IP phone and Ethernet switch As low a broadcast rate as possible 500 s recommended max 1000 s absolute max Ethernet Switches The following recommendations apply to Ethernet switches to optimize operation with Avaya IP telephones and other Avaya VoIP endpoints such as IP boards They are meant to provide the simplest configuration by removing unnecessary features Enable Spanning Tree fast start feature or disable Spanning Tree on host ports The Spanning Tree Protocol STP is a layer 2 L2 protocol used to prevent loops when multiple L2 network devices are connected together When a device is first connected or re connected to a port running STP the port takes approximately 50 seconds to cycle through the Listening Learning and Forwarding states KW Avaya IP Telephony Implementation Guide 16 This 50 second delay is not necessary and not desired on ports connected to IP hosts non network devices Enable a fast start feature on these ports to put them into the Forwarding state almost imme
120. ol remote port networks you would guarantee 128Kbps 69 3Kbps 64Kbps for IPSI signaling bandwidth across the WAN link A standby IPSI consumes an additional 2 4 Kbps bandwidth on the standby link KW Avaya IP Telephony Implementation Guide 79 The following table summarized IPSI signaling bandwidth requirements BHCC sd Ethernet PPP MLPPP Frame Relay SK 128Kbps 28K bps 128Kbps 128Kbps Cisco CBWFQ allows you to specify the exact amount of bandwidth to be allocated for a specific class of traffic Taking into account available bandwidth on the interface you can configure up to 64 classes and control distribution among them Bandwidth can be assigned a percentage of total link speed or in Kbps A FIFO queue is reserved for each class Optional WRED can selectively discard lower priority traffic when the interface begins to get congested Lower priority traffic should be guaranteed 25 of the available bandwidth on a WAN link Implement a QoS policies that provides A queue for IPSI traffic using for example DSCP 36 AF42 for IPSI signaling traffic IPSI signaling can be assigned another DCSP value but must guarantee bandwidth to minimize Packet Loss Expedited Forwarding DSCP 46 like behavior for the real time voice Call Admission Control CAC can be used to limit VoIP bandwidth Assured Forwarding AF31 like behavior for H 323 Call Signaling Traffic Note DSCP 34 AF41 is reserved for Video in
121. ommended for use across limited bandwidth WAN links Notice that two thirds of the packet is consumed by overhead IP UDP and RTP and only one third 1s used by the actual audio IP Header UDP Hdr RTP Header 20ms of G 729 Audio 20 B 8 B 12 B 20B It is important to understand that all 20 ms G 729 audio packets regardless of the vendor are constructed like this Not only is the structure of the packet the same but the method of encoding and decoding the audio itself is also the same This sameness is what allows an Avaya IP phone to communicate directly with a Cisco IP phone or any other IP phone when using matching codecs The packets from the application perspective are identical Network Perspective RTP header compression is a mechanism employed by routers to reduce the 40 bytes of protocol overhead to approximately 2 to 4 bytes 7 p 1 2 p 5 14 Cisco routers employ this mechanism as does the Avaya X330WAN router which is a module for the P330 chassis RTP header compression can drastically reduce the VoIP bandwidth consumption on a WAN link when using 20 ms G 729 audio When the combined 40 byte header is reduced to 4 bytes the total IP packet size is reduced by 60 from 60 bytes to 24 bytes This equates to reducing the total VoIP WAN bandwidth consumption by roughly half and it applies to all 20 ms G 729 audio packets regardless of the vendor Customers who deploy routers capable of this feature may be able to benefit from it Ho
122. on s IP address does not fall into any of the ranges configured on this form the station is assigned to the same network region as the gatekeeper it registers with Whether by assignment on this form or by inheritance it is very important to assign IP stations to the proper network region To understand how these methods of network region assignment affect the station see the Avaya Communication Manager Network Region Configuration Guide at www avaya com The VLAN column is used to send a VID to IP phones This field should only be used if DHCP option 176 is not available If such is the case then two rows are required on this form one row for the data VLAN through which the phone passes and another row for the voice VLAN on which the phone finally resides with both rows containing the voice VID The resulting functionality would be as follows IP phone boots and obtains address on data VLAN KW Avaya IP Telephony Implementation Guide 40 IP phone registers with Communication Manager from data VLAN ip network map shows phone assigned to a specific network region on a specific voice VLAN Communication Manager directs phone to that voice VLAN IP phone releases data VLAN address and obtains address on voice VLAN IP phone registers with Communication Manager from voice VLAN ip network map shows phone assigned to a specific network region on a specific voice VLAN Communication Manager directs phone to that voice VL
123. operating at 10 half until the problem can be resolved IP Server Interface IPSI Board The IP Server Interface IPSI board is installed in a G650 G600 MCC1 or SCC1 port network and it is the port network s interface to communicate with the call server s Most of the programming for an IPSI board is done on the SAT ipserver interface form which has commands change ipserver interface display ipserver interface and list ipserver interface If IP Control is y the board is acting as an IPSI otherwise n it is acting as a tone clock The n option is primarily used for migrating a non IPSI port network to an IPSI port network Ignore Connectivity in Server Arbitration has to do with whether or not connectivity to this IPSI is factored into the decision to interchange S87xx servers In most cases this is set to n but in rare cases it could be set to y for IPSIs in remote locations with poor network connectivity back to the servers The intent would be to avoid server interchanges caused by frequent and inconsistent loss of communication to this IPSI Location is the board slot Host is the board s static IP address if configured manually or the hostname if the address was obtained via DHCP DHCP ID 1s the hostname Socket Encryption if the parameter 1s present allows the control link between the IPSI and call server to be encrypted When QoS is enabled the 802 1p and DiffServ parameters contain the values to
124. ous adjuncts and SAT administrative access via TCP IP Each connection for one of these functions requires at least one TCP socket on the C LAN board The C LAN board can support over 400 sockets under light usage conditions However with heavier usage comes greater load on the CLAN and worse performance Furthermore regardless of usage it is highly discouraged to operate the C LAN near maximum capacity in a production environment The following conservative recommendations are offered for typical environments and can vary based on usage levels Adjuncts such as CMS CDR AUDIX Messaging System and others should be placed on separate C LAN boards that are not used for call signaling or media gateway control signaling This 1s KW Avaya IP Telephony Implementation Guide 32 common practice primarily due to business impact of the adjunct and the consequent need to isolate the adjunct as well as to quickly troubleshoot any problems related to the adjunct Ina typical call center environment design for a normal operating load of 200 250 IP stations plus 6 media gateways per C LAN Ina typical non call center business environment design for a normal operating load of 250 300 IP stations plus 8 media gateways per C LAN The number of signaling groups IP trunks per CLAN depends greatly on the configuration and usage of each signaling group Configuring the signaling group to have calls share IP signaling connection an option betwe
125. p Q and DSCP configurations are very similar The G350 does not have an Octaplane interface G250 Media Gateway The G250 is very similar to the G350 but with less capacity 10 users max The fundamental configuration requirements for the G250 are essentially the same as for the G350 as described above Like the G700 and G350 the G250 is capable of housing an S8300 server in ICC or LSP mode In addition unlike the other two gateways the G250 can act as a survivable H 323 gatekeeper This feature is called Standard Local Survivability SLS and it allows the G250 to be a call server with very limited features when communication to the primary call server is lost For simplicity SLS can be thought of as an integrated LSP with very limited features The details of SLS and its configuration are not covered in this document KW Avaya IP Telephony Implementation Guide 31 General Guidelines Related to Gateways The MGC List Transition Point and Primary Search Time must be configured properly on all gateways These are the parameters that determine which devices are primary controllers which is the LSP and when to fail over to the LSP There may be undesirable behaviors if these parameters are not configured properly such as a gateway registering with an LSP when a primary controller is available a gateway registering with an LSP too soon after an outage and different gateways at the same location registering with the LSP at differ
126. pendices A and B for more information and updates regarding auxiliaryvlan and trunking Properly configure 802 1Q trunking on Cisco switches If trunking 1s required on a Cisco CatOS switch connected to an Avaya device enable it for 802 1Q encapsulation in the nonegotiate mode set trunk lt mod port gt nonegotiate dotlq This causes the port to become a plain 802 1Q trunk port with no Cisco auto negotiation features When trunking is not required explicitly disable it as the default is to auto negotiate trunking Speed Duplex One major issue with Ethernet connectivity is proper configuration of speed and duplex There is a significant amount of misunderstanding in the industry as a whole regarding the auto negotiation standard The speed can be sensed but the duplex setting is negotiated This means that if a device with fixed speed and duplex is connected to a device in auto negotiation mode the auto negotiating device can sense the other device s speed and match it But the auto negotiating device cannot sense the other device s duplex setting the duplex setting is negotiated Therefore the auto negotiating device always goes to half duplex in this scenario The following table is provided as a quick reference for how speed and duplex settings are determined and typically configured It 1s imperative that the speed and duplex settings be configured properly Devicel Device2 Result Configuration Configuration 100 full exp
127. phones such as the 4606 4612 4624 and 4630 contain a 10 100 hub The last and best firmware for these models is version 1 8 3 legacy models cannot accept firmware newer than this The integrated hub in the legacy IP Telephones operate at 10 mbps or 100 mbps half duplex When connected to an Ethernet switch port that is configured to auto negotiate the Ethernet switch port stabilizes at 100 half The exception to this is if a personal computer is attached to the telephone that is capable of only 10 mbps In this case all three devices stabilize at 10 half If no personal computer is to be attached to the telephone or if the attached computer will always be capable of 100 mbps operation it is good practice to lock down the Ethernet switch to 100 half If a personal computer might be attached to the telephone and there 1s a chance that the computer might have a 10 mbps NIC leave the Ethernet switch port in auto negotiate mode These older telephones however cannot operate in full duplex mode The term current models in this document refers to the 4620 and 4610 and models containing the SW designation Current model telephones have an internal Ethernet switch that allows the telephone and a PC to share the same LAN connection if appropriate Thus SW models do not need or work with the 30A applicable to the 4612 4624 4630 only switched hub interface The exception to this exception 1s the 4620 both the 4620 and 4620SW contain an Eth
128. phones typically come from the factory with outdated code In addition the 46xxupgrade scr script may instruct the phone to download the 46xxsettings scr or txt file which is an optional method of sending configurations to the phone Note that there is a loading period after each bin code is received for the first time Note also that the file names are case sensitive on some servers Unix Linux and not on others Microsoft Ext and Password prompts The phone prompts for the extension and password if there are no previously stored values Registration with gatekeeper The phone registers with a gatekeeper C LAN or S8300 after the extension and password are entered This registration happens very quickly and does not show up on KW Avaya IP Telephony Implementation Guide 48 the display However the following packets are exchanged RAS Gatekeeper Request GRQ from phone to gatekeeper RAS Gatekeeper Confirm GCF from gatekeeper to phone RAS Registration Request RRQ from phone to gatekeeper not necessarily the same one GRQ was sent to RAS Registration Confirm RCF from gatekeeper to phone H 225 call signaling connection The phone opens a TCP session with the gatekeeper and sends an H 225 Setup message which is answered with H 225 Call Proceeding and H 225 Connect messages from the gatekeeper This call signaling session remains up throughout the registration During idle periods the phone main
129. public switched telephone network Another benefit is the potential to more tightly integrate data and voice applications Because they use open programming standards Avaya MultiVantage products make it easier for developers to create and for companies to implement applications that combine the power of voice and data in such areas as customer relationship management CRM and unified communications A converged multi service network can make such applications available to every employee These benefits do not come free however Voice and data communications place distinctly different demands on the network Voice and video are real time communications that require immediate transmission Data does not Performance characteristics that work fine for data can produce entirely unsatisfactory results for voice or video So networks that transmit all three must be managed to meet the disparate requirements of data and voice video Network managers are implementing a range of techniques to help ensure their converged networks meet performance standards for all three payloads voice video and data These techniques include the strategic placement of VLANs and the use of Class of Service CoS packet marking and Quality of Service QoS network mechanisms For an overview of IP telephony issues and networking requirements see the Avaya IP Voice Quality Network Requirements white paper Professional consulting services are available through the
130. r method are merged into one list IP telephone 2 0 and later maintain the two lists separately with only one list active at any given time During boot up the phone uses the list obtained from option 176 After registration the phone uses the Alternate Gatekeeper List received in the RCF When a phone 1s logged off but not rebooted it reverts back to the list obtained from option 176 The option 176 GK list is recommended as opposed to manual entry or a single GK address in option 176 because the RCF list is received after registration If the phone only knows of one GK at boot up and that GK is out of service the phone cannot register and hence cannot get an RCF The Alternate Gatekeeper List sent in the RCF follows a specific algorithm When an IP phone registers and its network region is specified in the ip network map form the call server delivers a list of all gatekeepers in that region plus directly connected regions specified in the ip network region form If an IP phone s network region is not administered in the ip network map form it inherits the region of the gatekeeper that receives the registration and the call server delivers a list of all gatekeepers only in that region The ip interface form includes an administered Gatekeeper Priority Value between 1 and 9 where 1 is the highest priority This value is used to build the RCF Alternate Gatekeeper list delivered in the RCF message If multiple gatekeepers have the same p
131. r IP ToS field This original scheme was not widely used and the IETF came up with a new marking method for IP called Differentiated Services Code Points DSCP RFC 2474 2475 DSCP utilizes the first six bits of the ToS field and ranges in value from 0 to 63 The following figure shows the original ToS scheme and DSCP in relation to the eight bits of the ToS field o L2 3 4 5 6 7 pC its 0 KW Avaya IP Telephony Implementation Guide 23 Figure 16 Compare DSCP w original ToS Ideally any DSCP value would map directly to a Precedence ToS combination of the original scheme This is not always the case however and it can cause problems on some legacy devices as explained in the following paragraph On any device new or old having a non zero value in the ToS field cannot hurt if the device is not configured to examine the ToS field The problems arise on some legacy devices when the ToS field is examined either by default or by enabling QoS These legacy devices network and endpoint may contain code that only implemented the IP Precedence portion of the original ToS scheme with the remaining bits defaulted to zeros This means that only DSCP values divisible by 8 XX X000 can map to the original ToS scheme For example if an endpoint is marking with DSCP 40 a legacy network device can be configured to look for IP Precedence 5 because both values show up as 10100000 in the ToS field However a DSCP of 46 101110 ca
132. r end of the IP trunk The IP trunk is not tied to any given MedPro MR320 or VoIP module As long as there is at least one MedPro MR320 or VoIP module at each end with available DSP resources the IP trunk is unaffected by MedPro MR320 or VoIP module failures Ifall usable Medpros or VoIP modules fail the IP trunk s trunk group members go out of service but the signaling group stays in service and can be used to send messages between the two systems This essentially results in a bypass condition where the TDM trunk is utilized Q7 How is call processing affected in general by a C LAN outage When configured properly the stations and media gateways have a list of alternate gatekeepers They discover ifa C LAN they are registered with has gone down and re home to a different C LAN Ifthe C LAN failure occurs during an active call the H 323 and H 248 link bounce recovery features preserve active calls on stations and media gateways respectively Q8 How is call processing affected in general by a MedPro MR320 or VoIP module outage The call server knows when a MedPro MR320 or VoIP module has gone out of service and stops directing calls to that device As long as there are sufficient MedPros MR320s or VoIP modules to compensate for the outage there is no adverse effect If there is an outage during an active call and that call is going through the affected MedPro MR320 or VoIP module that call loses audio Avaya is studying the concept of red
133. r problem The show port lt mod port gt command on Catalyst switches gives this information The Avaya P550 commands are show port status lt mod port gt show port counters lt mod port gt and show ethernet counters lt mod port gt The Avaya P330 switch command is show rmon statistics lt mod port gt 2 2 Bandwidth Considerations Calculation Many VoIP bandwidth calculation tools are available as well as pre calculated tables Rather than presenting a table the following information is provided to help the administrator make an informed bandwidth calculation The per call rates for G 711 G 726 and G 729 are provided under the Ethernet Overhead and WAN Overhead headings below and all calculations are for the recommended voice packet size of 20ms Voice payload and codec selection The G 711 codec payload rate is 64000bps Since the audio is encapsulated in 10 ms frames and there are 100 of these frames in a second 100 10ms Is each frame contains 640 bits 64000 100 or 80 bytes of voice payload The G 726 codec payload rate is 32000bps and the G 729 codec payload rate is 8000bps This equates to 320 bits or 40 bytes and 80 bits or 10 bytes per 10 ms frame respectively Voice Payload 1 frame 10ms 2 frames 20ms 3 frames 30ms 4 frames 40ms G 711 160 B 240 B 320B G 726 120 B 160 B G 729 Table 2 Voice payload vs number of frames Packet size and packet rate Because the voice payload
134. rate must remain constant the number of voice frames per packet packet size determines the packet rate As the number of frames per packet increases the number of packets per second decreases to maintain a steady rate of 100 voice frames per second Packet Rate Codec Payload 1 frame packet 2 frames packet 3 frames packet 4 frames packet Rate 10ms 20ms 30ms 40ms G 711 64000bps 100pps SOpps 33pps 25pps G 726 32000bps 100pps SOpps 33pps 25pps G 729 8000bps 100pps SOpps 33pps 25pps Table 3 Packet rate vs packet size IP UDP RTP overhead Each voice packet inherits a fixed overhead of 40 bytes IP UDP RTP Voice Payload 20 B 8 B 12 B Variable KW Avaya IP Telephony Implementation Guide 18 Figure 11 IP UDP RTP overhead To this point the calculation is simple Add up the voice payload and overhead per packet and multiply by the number of packets per second Here are the calculations for a G 711 and a G 729 call both using 20 ms packets Remember that there are 8 bits per byte G 711 160B voice payload 40B overhead packet 8b B 50 packets s 80kbps G 726 80B Voice payload 40B overhead packet 8b B 50 packets s 48kbps G 729 20B voice payload 40B overhead packet 8b B 50 packets s 24kbps The calculations above do not include the L2 encapsulation overhead L2 overhead must be added to the bandwidth calculation and this varies with the protocol being used Ethernet PPP HDLC ATM Frame Relay etc
135. reason the phone attempts other gatekeepers in the gatekeeper list The following hypothetical network diagram and the accompanying instructions explain how gatekeeper lists should be administered on DHCP servers KW Avaya IP Telephony Implementation Guide 53 SDR gt as DHCP Server w scopes for v80 90 Janik Network Region 2 Access Access Switches Switches Voice v80 Voice v90 DHCP Server w scopes for v10 40 Network Region 1 g700 with Core Core 8300 LSP i wo e eas tase aS N pane of lt a a v80 get info v90 get info Distribution Distribution Distribution Distribution from DHCP from DHCP Switch Switch Switch Re CLANA v80 scope v90 scope lt O Access Access Access Access Switches Switches Switches Switches Voice v10 Voice v20 Voice v30 Voice v40 P VA phones on phones on phones on phones on phones on phones on v10 get info v20 get info v30 get info v40 get info from DHCP from DHCP from DHCP from DHCP v10 scope v20 scope v30 scope v40 scope re a Figure 18 Hypothetical converged network Main Site The converged network depicted in the figure above could be an entire network or a portion of a much larger network The main site is implemented in a core distribution access architecture common to many enterprise networks The IP phones are scattered across various voice VLANs but the phones all belong to the same Communication Manager network region because they use the
136. returns to no tagging back to the data VLAN The idea behind going back to the data VLAN after some time is that the phone may have changed ports and be on one with a different voice VLAN In such a case the phone would have to start over and be directed to the proper voice VLAN The idea behind marking VLAN 25 as invalid in the previous scenario is that if the phone hasn t changed ports it is preferable to operate on the data VLAN than to be KW Avaya IP Telephony Implementation Guide 52 sent to a bad voice VLAN in a continuous loop For cases where it is not preferable to operate on the data VLAN the option VLANTEST 0 was added as of legacy phone firmware 1 8 2 and current phone firmware 2 0 1 This instructs the phone to permanently remain on the previously known voice VLAN Once the phone accepts VLANTEST 0 or marks a VLAN as invalid the only way to clear out this state is by manually resetting the values via the Hold RESET menu Note DHCP option 176 is the preferred method for directing IP phones to the voice VLAN The method described previously using the VLAN field of the ip network map form is an alternative if DHCP option 176 is not available The two methods should not be used simultaneously Phone firmware 2 4 1 changes the values and behavior of VLANTEST The value range increases from 0 999 16 65 minutes seconds to 0 172800 48 hours Also when the timer expires the voice VLAN 1s NOT marked as invalid Instea
137. riority value then the gatekeeper list is based on socket load per gatekeeper within the same priority KW Avaya IP Telephony Implementation Guide 55 As of Avaya Communication Manager 1 3 the addresses of the LSPs administered on the ip network region form in the same network region as the IP phone are also sent in the RCF As of Communication Manager 2 0 in addition to the LSPs the address of the Survivable GK Node Name administered on the station form is also sent in the RCF The combination of Communication Manager 2 x and IP telephone 2 x facilitates a distinction between primary and secondary gatekeepers in the Alternate Gatekeeper List During recovery after an outage the primary gatekeepers are attempted first for a period of time called the H 323 Primary Search Time specified in the system parameters ip options form After this search time expires the secondary gatekeepers LSPs and the Survivable GK are also included in the search For a more detailed discussion see the H 323 Link Bounce section of the Avaya Communication Manager Network Region Configuration Guide at www avaya com Verifying the Gatekeeper Lists The table below gives a summary of how to view the gatekeeper and gatekeeper list in use Phone state Registered phone Phone logged off via Hold Method LOGOFF keypad command 2 1 and later shows gatekeeper 2 1 and later shows list received from option 176 or Alternate Gatekeeper Lis
138. roduction environment KW Avaya IP Telephony Implementation Guide 73 Other Examples Example 3 Suppose that C LANs 192 168 1 10 and 11 cannot mark their traffic pre Communication Manager system This set of configurations is applied only to the left router access list 101 permit ip host 192 168 1 10 192 168 2 0 0 0 0 255 access list 101 permit ip host 192 168 1 11 192 168 2 0 0 0 0 255 Access list 101 permits any IP traffic from the two C LANs to the 192 168 2 0 24 network There is an implicit deny any at the end of this access list class map match any untaggedVoIP create a class map called untaggedVoIP match access group 101 packets matching access list 101 are in the class untaggedVoIP policy map setDSCP create a policy map called setDSCP class untaggedVoIP for all packets in the class untaggedVolIP set the DSCP to 46 set ip dscp 46 interface Ethernet 0 0 service policy input setDSCP apply the setDSCP policy inbound on this interface Now the C LAN traffic is marked with DSCP 46 as in example 2 and the example 2 configurations must be applied to both routers Example 4 This is the same as example 2 but with more restrictions on the traffic In this example DSCP 46 is used throughout to simplify the access list A somewhat matching set of configurations is applied to both routers access list 101 permit ip 192 168 1 0 0 0 0 255 192 168 2 0 0 0 0 255 dscp 46 left router access list 101 permit ip 192 168 2 0 0 0 0 255 192 168
139. rver Interface Board For the 802 1p priority from the SAT form to be applied to the S87xx S8500 server s L2 tagging must be enabled on the appropriate server interfaces via the Configure Server Configure Interfaces web admin screen The interfaces that communicate with IPSI boards have this option and the others do not The VLAN ID is always 0 for the S87xx S8500 servers follow the instructions in section 2 3 heading Rules for 802 1p Q Tagging 3 2 8300 Server The S8300 is a Red Hat Linux server platform similar to the S87xx S8500 but on a compact media module that fits into a G700 G350 G250 gateway always in media module slot 1 The 8300 is similar to the S87xx S8500 in many ways It is configured and managed via the same web interfaces and as with the other servers there 1s little or no need for a customer to access the Linux shell In a G700 the S8300 must have an IP address on the same IP subnet as the MGP with the same mask and default gateway see G700 section below This is because all media module slots in a G700 inherit the VLAN of the MQP and therefore all VoIP media modules and the S8300 must be on the same IP subnet as the MGP In a G350 G250 a VLAN must be designated as the ICC VLAN and the S8300 must have an IP address on the IP subnet pertaining to that VLAN see G350 section below An 8300 server can be in one of two modes internal call controller ICC or local survivable processor LSP In ICC mo
140. s There is also an Octaplane slot on the back of the chassis just like the P330 For practical purposes the L2 switching portion of the G700 is equivalent to a 2 port P330 switch which has a CLI similar to Cisco s CatOS and is configured using various set commands Three components of the P330 should be configured the inband management interface the default route and the switch itself The inband management interface 1s displayed and configured using the commands show interface inband and set interface inband respectively The inband interface requires a VLAN an IP address and a mask The VLAN can be any of the VLANs active on the P330 and the IP address and mask must correspond to the IP subnet associated with that VLAN Once configured the inband interface should be thought of as a host attached to the P330 This may seem non intuitive because the inband interface is the P330 and the way to administer the P330 remotely However like most L2 switch management interfaces the inband interface 1s associated with a specific VLAN As such it is accessed just like any other host attached to the switch on a given VLAN either directly from another host on the same VLAN subnet or by routing to it from a host on a different VLAN subnet Many mistakenly think that any host attached to the P330 should be able to access the inband interface directly and this is not necessarily true Hosts on different VLANs subnets must route to the inband managem
141. same codec set share the same audio characteristics and use the same resources specified by a network region Network region has four C LANSs scattered across four distribution switches but there could be more depending on the number of IP telephones The fact that there are four C LANs and four voice VLANs is purely coincidental Suppose for whatever reason that a large number of IP phones are rebooted at once Which gatekeeper s will they contact first The correct answer is that they should contact all the gatekeepers in a distributed fashion All the phones should not bombard the same gatekeeper at once with GRQs There are various ways to configure the gatekeeper lists and the following is possibly the simplest vl0 scope MCIPADD clanladdr clan2addr clan3addr clan4addr v20 scope MCIPADD clan2addr clan3addr clan4addr clanladdr v30 scope MCIPADD clan3addr clan4addr clanladdr clan2addr v40 scope MCIPADD clan4addr clanladdr clan2addr clan3addr Based on how this particular network is implemented here is another alternative vl0 scope MCIPADD clanladdr clan2addr clan3addr clan4addr v20 scope MCIPADD clan2addr clanladdr clan4addr clan3addr v30 scope MCIPADD clan3addr clan4addr clanladdr clan2addr v40 scope MCIPADD clan4addr clan3addr clan2addr clanladdr Regardless of how the lists are administered the principle is important DHCP
142. se choice If voice quality is not acceptable or if QoS is desired for contingencies such as unexpected traffic storms the best place to begin implementing QoS is on the WAN link s Then QoS can be implemented on the LAN segments as necessary One caution to keep in mind about QoS is regarding the processor load on network devices Simple routing and switching technologies have been around for many years and have advanced significantly Packet forwarding at L2 and L3 is commonly done in hardware Cisco calls this fast switching 2 p 5 18 switching being used as a generic term here without heavy processor intervention When selection criteria such as QoS and other policies are added to the routing and switching function it inherently requires more processing resources from the network device Many of the newer devices can handle this additional processing in hardware resulting in maintained speed without a significant processor burden However to implement QoS some devices must take a hardware function and move it to software Cisco calls this process switching 2 p 5 18 Process switching not only reduces the speed of packet forwarding but it also adds a processor penalty that can be significant This can result in an overall performance degradation from the network device and even device failure Each network device must be examined individually to determine if enabling QoS will reduce its overall effectiveness by moving a hardware fun
143. se requires too many renewals which not only taxes the DHCP server but can also disrupt service to the IP phones if renewals cannot be accomplished for whatever reason On the other hand too long a lease can result in IP address exhaustion if hosts are unplugged from the network without properly shutting them down to invoke a release of the IP address lease Additional Script and Firmware Download Methods Beginning with Avaya IP Telephone Release 2 2 Avaya IP phones can download scripts and firmware from web servers using the HTTP or TLS HTTPS protocols in addition to TFTP Preliminary testing at Avaya Labs indicates that HTTP servers can support more simultaneous downloads than TFTP servers suggesting that HTTP TLS are better suited for large IP telephone deployments than TFTP To specify TLS or HTTP script firmware downloads in option 176 of the DHCP scope apply the TLSSRVR for TLS or HTTPSRVR for HTTP parameter in lieu of TFTPSRVR If TLSSRVR HTTPSRVR and TFTPSRVR are all set the phone will attempt to download firmware using TLS first on TCP port 411 then HTTP on TCP port 81 then HTTP on TCP port 80 then TFTP on UDP port 69 Note Avaya IP telephones only establish encrypted TLS connections with servers using an Avaya signed digital certificate ie an Avaya S8300 or S8500 Media Server Boot up Sequence The following are key boot up events listed in order which may help to verify proper operation of the IP phone This l
144. side reverts back to auto negotiate for any reason it will show the negotiated speed duplex to be 100 half which is a duplex mismatch and must be corrected Following the instructions in section 2 1 heading Speed Duplex examine the error counters on the Ethernet switch port and verify that the link is healthy no errors S 87xx S8500 802 1p Q and DSCP On a Multi Connect system the port network control traffic between the S87xx S8500 server s and IPSI boards traverses a closed control IP network On this network there is no need to configure QoS because all traffic is port network control traffic and has equal priority QoS is required when there is the potential for contention for resources such as bandwidth queue space and processing power between various classes of traffic This does not apply on the control IP network On an IP Connect system the port network control traffic traverses the enterprise IP network which services various classes of traffic If QoS is desired and properly configured on this network it may be necessary to have the S87xx S8500 server s tag mark the port network control traffic This is only required on the interfaces that communicate with IPSI boards as they are the only ones that participate in real time IP telephony Traffic is tagged marked from these interfaces on a per destination basis for each IPSI board as administered on the SAT ipserver interface form see section 3 4 heading IP Se
145. solete Obsolete n a 2 x and later w configurable configurable varies CM2 x and later Table 9 TCP and RAS keepalive matrix 4 2 Connecting a PC to the Phone On the back of the phone the port with the icon that looks like a terminal is the user port The port with the icon that looks like a network jack is the uplink port which connects to the Ethernet switch Use discretion when connecting a PC to the phone and remember that its primary function is not that of an enterprise network device For example do not connect an enterprise server to the phone Such high traffic servers require their own separate connections to the enterprise Ethernet switch Also do not connect a PC to the phone with a 10M uplink to the network The phone itself operates well at 10M but with a PC attached the two should operate at 100M IP Phone and Attached PC on Same VLAN There are three variations of attaching a PC to the phone and the first two involve having both the phone and the PC on the same VLAN which is the port native VLAN refer to Appendix A for a primer on VLANs In the first scenario traffic from both the phone and the PC have no CoS tagging In this case no special configurations are necessary Simply attach the phone to an access port one with only the port native VLAN configured and attach the PC to the phone The second scenario is similar to the first except that traffic from the phone 1s marked with L2 and or L3 priority while re
146. ss should not be greater than 3 The IPSI circuit pack provides enterprises with the capability to IP connect Port Networks over LAN WAN links in simplex and high availability configurations Call signaling and system maintenance traffic is passed between S87XX S8500 call control servers and the IPSI circuit packs in a port network The call signaling traffic is encapsulated AVAYA proprietary CCMS Control Channel Message Set messages inside TCP IP packets The CCMS messages are H 323 H 225 and Q 931 messages used for registration of IP endpoints to setup and teardown calls periodic testing of the hardware and keep alive messages for IPSI connected port networks These CCMS messages are critical to the stability of a port network and delivery of CCMS messages must be guaranteed The following table and graph displays IPSI call signaling traffic for varying Busy Hour Call Completion rates BHCC BHCC IPSI bandwidth is based on 150 IP endpoints originating and answering 10 second duration ISDN trunk calls within a port network The simulated call scenario is a general business case The common defaults for station traffic usage in a general business scenario are light traffic moderate traffic and heavy traffic Per PN full ome socond E O a E A E Moderate Traffic OIK S S BBE B Heavy Traffic KW Avaya IP Telephony Implementation Guide 78 IPSI Call Signaling Packet Traffic 100 80 60 40 20 Kbps 1K 2 5K 5K 7 5K
147. ss tonya coseanadiaueuetovaians A aes 48 Call S126 1 rte ete eee Oe rarer PO errr Ire ODOT TTS re PT 49 Keepalive MechanisMs sournaru ena ar N AOA AOTEA 49 Avaya IP Telephony Implementation Guide 5 A Deca Ome Cine a WO tne PHO Me vse sec cccsrea wie E ENAA AE OAE TE 51 IP Phonesand Attached PC on Same VLAN sscisesesssceiescscnescnavereushviustemietvsocacasd neva rhashieuatetias 51 IP Phone and Attached PC on Different VLANS accuracies 52 4 3 4 Gatekeeper Lists and DHCP Option 17 6 wissescwsses no sedcetevnentaavies e A A E E TEE TRR 53 MaE SIG a taa taster age caveat tetae fasta agian arediet anata caster agen csiaeteasanes 54 Bianchi UC aes acne iui iacacr oie ecdaneunauua iuneaiaauantaaal ues vancuasoun ecinuen cata a vuatueaauoeeen a muanoeatniaaus 55 Two Methods or Receiving the Gatekeeper List cers access cccece nacetenectn ee anaes eee 55 Veriyi the Gatekeeper LiStSransssrcecierii na N S 56 APPENA AA NV LA N PE a E A E ule uaaustmiearesteaietues 57 Appendix Bi CISH Awo DISCOVER T 62 Appendix C RIP Header ConmipressiOn enano a E A 65 Appendix D ACcess ListOmide Hnes enasna r a N 67 Appendix E Common IPF COmmMINd Ssa A E N oes 69 Appendix F Sample O0S Contieuraions sorsia nia dentedan guacatoedechasausianeeceataienons 71 Appendix G IP Trunk Bypass TDM Fallback Q amp A ccccccccceesssssessssssnnnseeeeeeeeeeesseesssesesssssennnaaes 75 Appendix H IPSI Signaling Bandwidth Requirements cccccccccessesseeessennneeeeeeeeeeee
148. t In this case all but the native VLAN should be cleared off the trunk Would not accept VID zero for the native VLAN Opened a case with Cisco TAC and TAC engineer said it was a hardware problem in the 4000 Catalyst 4000 w Bug ID is CSCdr06231 Workaround is to enable 802 1Q trunking and tag CatOS 6 3 3 with native VID instead of zero Again clear all but the native VLAN off the trunk IOS 12 0 5 WC2 disabled on the port documentation or call TAC KW Avaya IP Telephony Implementation Guide 58 Note that setting a L2 priority is only useful if QoS is enabled on the Eth switch Otherwise the priority tagged frames are treated no differently than clear frames Sample Multi VLAN Scenario for Avaya P330 Code 3 2 8 and Cisco CatOS and IOS Here is a sample multi VLAN scenario Suppose there is a Cisco router connected to a P330 switch that contains two VLANs one for the VoIP devices and one for the PCs To conserve ports and cabling the PCs are connected to the phones and the phones are connected to the P330 switch C LAN MedPro vlan 10 vlan 10 192 168 10 1 192 168 10 2 vlan 10 0 1 1 2 1 3 192 168 10 7 1 1 1 5 a Avaya Iiis IP Phone Cisco Router 192 168 1 254 192 168 10 254 Cajun P330 1 12 vlan 1 vlan 1 fle DHCP Server PC TFTP Server Cisco Router configuration interface FastEthernet0 1 description 802 1Q trunk interface interface FastEtherne
149. t manually configured received in RCF message gatekeeper which is the list in use 2 0 1 shows gatekeeper list 2 0 1 shows gatekeeper list received from option 176 or MIB Object ID received from option 176 manually configured 1 3 6 1 4 1 6889 2 69 1 1 3 or manually configured gatekeeper endptMCIPADD gatekeeper even though the list in use is the Alternate Gatekeeper List from RCF 1 8 x shows combined list from RCF and option 176 or 1 8 x shows combined list combined RCF list and from RCF and option 176 manually configured or combined RCF list and gatekeeper manually configured gatekeeper MIB Object ID Shows gatekeeper to which Shows gatekeeper to which phone 1 3 6 1 4 1 6889 2 69 1 1 4 phone is currently registered was last registered endptMCIPINUSE MIB Object ID 2 2 and later shows the 2 2 and later shows the alternate 1 3 6 1 4 1 6889 2 69 1 1 4 28 alternate gatekeeper list gatekeeper list received from CM in endptRASGKkList received from CM in the RCF the RCF Hold ADDR keypad menu Shows gatekeeper to which N A phone is currently registered KW Avaya IP Telephony Implementation Guide 56 Appendix A VLAN Primer This appendix is primarily concerned with configurations that require the Avaya IP Telephone to connect to an Ethernet switch Eth switch port configured with multiple VLANs the IP phone on one VLAN and a PC connected to the phone on a separate VLAN Three sets of configurations are give
150. t values 6 6 are used Note The L3 priority values DSCP are received from the call server as configured on the SAT ip network region form The reason L3 values are received from the call server and L2 values are not is because an IP phone accepts all L2 values from one source The preferred and recommended method is via DHCP option 176 An alternative method is described in section 3 5 heading i1p network map which utilizes the L2 values administered on the SAT ip network region form An administrator must create option 176 on the DHCP server and administer a properly formatted string with the appropriate values Option 176 could be applied globally or on a per scope basis The recommendation is to configure option 176 on a per scope basis because the values themselves or the order of the values could change on a per scope basis As part of the DHCP process at boot up the IP telephone requests option 176 from the DHCP server KW Avaya IP Telephony Implementation Guide 47 DHCP Lease Duration A DHCP server gives out an IP address with a finite or infinite lease and the Avaya recommended lease duration for IP phones is 2 to 4 weeks The DHCP specification calls for the client to renew the lease at determined intervals typically beginning at half life of the lease If the first renewal attempt fails there are allowances in the specification for further renewal attempts dependent on the length of the lease Too short a lea
151. t0 1 1 encapsulation dotlq 1 ip address 192 168 1 254 255 255 255 0 interface FastEthernet0 1 10 encapsulation dotlq 10 ip address 192 168 10 254 255 255 255 0 ip helper address 192 168 1 100 To forward DHCP requests to the DHCP server All ports have port native VLAN 1 by default set port vlan binding mode 1 1 static Port in static binding mode by default but command shown set port static vlan 1 1 10 In addition to v1 v10 statically bound to port set trunk 1 1 dotlq Port connected to Cisco router is an 802 1Q trunk port set port spantree disable 1 1 Spanning Tree disabled at the port level set port vlan 10 Port native VLAN changed to 10 on this port set port spantree disable Spanning Tree disabled at the port level set port level 6 Port L2 802 1p priority set to 6 set port vlan 10 1 3 set port spantree disable 1 3 set port level 1 3 6 KW Avaya IP Telephony Implementation Guide 59 set port vlan binding mode 1 5 static Port in static binding mode by default but command shown set port static vlan 1 5 10 In addition to v1 v10 statically bound to port but not a trunk port set port spantree disable 1 5 Spanning Tree disabled at the port level Port 1 12 for the DHCP TFTP server already has port native VLAN 1 set port spantree disable 1 12 Spanning Tree disabled at the port level P330 C360 configuration bind to configured option All ports have port native VLAN 1 by default set vlan 1 vl configured set vl
152. tack might be required when the 16 port X330 Ethernet expansion module is used in the G700 and the hosts attached to that module communicate mostly to other hosts not on the G700 If the hosts on the expansion module are IP telephones a 100M uplink is sufficient But if PCs are attached to the phones and the PCs frequently communicate off the G700 a 100M uplink may not be sufficient G350 Media Gateway The G350 is similar to the G700 in many ways Therefore this section details the differences while referring to the G700 explanations for similarities Two significant differences between the G350 and G700 are capacity and architecture The G350 supports much fewer users 40 max than the G700 450max As such the G350 s internal VoIP module has only 32 audio resources as opposed to 64 in the G700 s internal VoIP module and in the external VoIP media module and MedPro board The G350 also cannot presently accept external VoIP modules The primary architectural difference between the G350 and G700 is that the G350 is an integrated platform The L2 switch MGP and internal VoIP module all share the same processing engine and same IP address In addition a L3 router is integrated into the G350 whereas the G700 can accept a L3 router as an expansion module The resulting G350 CLI has two components The L2 switch and MGP commands are practically the same as on the G700 using set commands similar to the P330 switch and Cisco s CatO
153. tains the session by sending TCP keepalives Phone is operational The administered display shows up on the phone and the extension LED illuminates on 4606 12 24 models Unregistration messages If the gatekeeper intentionally unregisters a set or if the set intentionally unregisters itself the message sent by either the gatekeeper or the set is a RAS Unregistration Request URQ with a reason code that is deciphered in the hex decode of most protocol analyzers The acknowledgment message is a RAS Unregistration Confirm UCF Call Sequence It is not feasible to give a standard packet by packet call sequence because of the many possible variations on any given call Instead a higher level description of the process is offered here Depending on which features are enabled and executed during a call the packet by packet sequence may vary but the fundamental functions described here apply overall All call signaling functions go through the gatekeeper either via the C LAN or natively S8300 and the gatekeeper dictates what the IP stations do during a call Calling phone contacts gatekeeper on already established call signaling session TCP 1720 gatekeeper port variable phone port There are some call signaling exchanges on this TCP session Calling phone establishes an audio stream with an audio resource MedPro MR320 board or VoIP module as directed by the gatekeeper Gatekeeper contacts called phone on alread
154. ters be set on the system parameters ip options form How do these settings affect the IP trunk bypass feature The system parameters ip options form is used to define the thresholds that trigger a fallback to a TDM trunk thus bypassing the IP trunk For this feature to work the Bypass if IP Threshold Exceeded parameter must be set to y in the signaling group form for an IP trunk and the correct route pattern must be administered Simply stated a near end MedPro MR320 monitors network performance by pinging the far end C LAN to measure network response against the configured thresholds One thing to note about the IP trunk bypass feature is that it is not fully supported on the 8300 media gateway platform The VoIP module in the G700 does not behave exactly like the MedPro MR320 board and it cannot perform the ping functions that a MedPro or MR320 performs The issues with an S8300 media gateway are discussed throughout this appendix When a high threshold is reached the signaling group goes into bypass state and a fallback TDM trunk is utilized When the corresponding low threshold is re established the signaling group comes back into service and the IP trunk is utilized Because networks and user preferences vary there is no single set of optimal thresholds This is a feature that must be tested and fine tuned with each implementation The parameters are as follows Roundtrip Propagation Delay ms High 400 500ms 1s a good star
155. th the S8300 LSP A TCP session is initiated from the primary server to the LSP TCP port 514 A second session is then initiated from the LSP to the primary server TCP port range 512 1023 Permit Permit This allows the Communication Manager 2 x primary server to synchronize translations with the 8300 LSP Permit LSP TCP 21873 Permit Primary server enterprise intfc TCP any This allows the Communication Manager 3 x primary server to synchronize translations with the 8300 LSP Permit LSP TCP 21874 Permit Primary server enterprise intfc TCP any This allows an administrator to log in via Avaya SA to a call server S87xx S8500 S8300 Permit TCP 5023 Permit TCP any This allows secure and unsecure web access to a call server S87xx S8500 S8300 The call server redirects unsecure sessions to https Permit S8xxx enterprise interface Permit Web admin station s Permit S8xxx enterprise interface Permit Web admin station s Optional services used by call server S87xx S8500 S8300 Permit DNS server s Permit S8xxx enterprise interface Permit NTP server s UDP 123 ntp Permit S8xxx enterprise interface UDP any H 248 signaling between G700 G350 G250 Media Gateway and S8300 or CLAN MG initiates session Permit S8300 or CLAN TCP 2945 Permit G700 G350 G250 TCP any H 248 encrypted signaling between G700 G350 G250 Media Gateway and S8300 or CLAN MG initiates session Permit TCP 1039 Permit TCP any Control network traffic and other tra
156. the Cisco AutoQoS model but can be assigned for IPSI traffic when video is not deployed IPSI traffic classification can be assigned on a S87XX S8500 via the change ipsi server interface command The set diffserv 36 CLI command is used to mark traffic from IPSI to Server when you login to the IPSI Use the show qos display the assigned marking It is important to note that H 248 Gateway Control traffic as well as Call Center configurations will have a greater signaling bandwidth requirement Consult your account team for additional traffic requirements References 1 Cisco Systems Inc Cisco IP Telephony Network Design Guide www cisco com Customer Order Number DOC 7811103 Copyright 2001 2 Cisco Systems Inc Cisco IP Telephony QoS Design Guide www cisco com Customer Order Number DOC 7811549 Copyright 2001 KW Avaya IP Telephony Implementation Guide 80 3 Cisco Systems Inc Configuring Compressed Real Time Protocol www cisco com July 2002 4 Cisco Systems Inc Troubleshooting Cisco Catalyst Switches to Network Interface Card NIC Compatibility Issues www cisco com July 2002 5 Cisco Systems Inc Understanding Compression Including cRTP and Quality of Service www cisco com July 2002 6 IEEE Inc 802 1Q IEEE Standard for Local and Metropolitan Area Networks Virtual Bridged Local Area Networks www iee org December 8 1998 7 IETF RFC 2508 Compressing
157. tically receives this value from the call server per the ip network region form This value could also be set manually on a per phone basis L3 signaling Layer 3 DSCP for signaling IP packets The phone automatically receives this value from the call server per the ip network region form This value could also be set manually on a per phone basis The manual menus are covered here for explanatory purposes However a better alternative is to use DHCP option 176 and the built in capabilities of the call server and IP telephone to automatically configure the phones As stated previously the call server sends the L3 priority values to the phones KW Avaya IP Telephony Implementation Guide 51 automatically per the values configured in the ip network region form The 802 1Q on off instruction VLAN ID and L2 priorities can be configured automatically using DHCP option 176 as described in section 4 1 heading DHCP Option 176 Here is what that string should look like for 1 8 and later phones see the appropriate LAN Administrator s Guide for previous phone releases MCIPADD adadr1 addr2 HTTPSRVR addr L2QVLAN 0 L2QAUD L2QSIG The LZ2QVLAN 0 parameter instructs the phone to enable 802 1 p Q tagging with VID 0 which means that the phone s traffic belongs on the port native VLAN The Ethernet switch port to which the phone is connected must be configured to accept 802 1Q tagging for this to work and the switch must i
158. ting point for this threshold Many users begin to notice performance degradation at around 200 250ms one way delay Roundtrip Propagation Delay ms Low 200 300ms 1s a good starting point for this threshold 100 150ms or less one way delay typically results in very acceptable audio quality Packet Loss High 7 10 is a good starting point for this threshold Avaya Labs testing has shown that audio quality is acceptable even with 5 packet loss Packet Loss Low 0 3 is a good starting point for this threshold Ping Test Interval sec This is the frequency at which pings are sent out The lower the interval the better for measuring network performance In loads prior to Avaya Communication Manager 2 1 the low limit is 10sec which is sufficient for detecting a network outage but not for measuring network performance As of Communication Manager 2 1 and MedPro firmware v70 the minimum ping test interval is Isec which is granular enough to gauge network performance 1 2 sec is a good starting point for this parameter Number of Pings per Measurement Interval This is the number of pings sent out before delay and loss are calculated 10 should be used here for a minimum ping test interval of 10sec which results in calculations every 100sec to detect a network outage As of Communication Manager 2 1 and MedPro firmware v70 20 to 30 pings at 1 second intervals results in calculations every 20 to 30 seconds which provides the
159. to implement QoS on a router than on an Ethernet switch Unlike Ethernet switches routers typically do not have a fixed number of queues Instead routers have various queuing mechanisms For example Cisco routers have standard first in first out queuing FIFO weighted fair queuing WFQ custom queuing CQ priority queuing PQ and low latency queuing LLQ LLQ is a combination of priority queuing and class based weighted fair queuing CBWFQ and it is Cisco s recommended queuing mechanism for real time applications such as VoIP Each queuing mechanism behaves differently and is configured differently but following a common sequence First the desired traffic must be classified using DSCP IP address TCP UDP port or protocol Then the traffic must be assigned to a queue in one of the queuing mechanisms Then the queuing mechanism must be applied to an interface 2 p 1 7 3 4 3 5 5 2 The interface itself may also require additional modifications independent of the queuing mechanism to make QoS work properly For example Cisco requires traffic shaping on Frame Relay and ATM links to help ensure that voice traffic is allotted the committed or guaranteed bandwidth see Traffic Shaping on Frame Relay Links below in this section Cisco also recommends link fragmentation and interleaving LFD on WAN links below 768kbps to reduce serialization delay Serialization delay is the delay incurred in encapsulating a L3 packet in a L2 frame and
160. transmitting it out the serial interface It increases with packet size but decreases with WAN link size The concern 1s that large low priority packets induce additional delay and jitter even with QoS enabled This is overcome by fragmenting the large low priority packets and interleaving them with the small high priority packets thus reducing the wait time for the high priority packets The following matrix is taken directly from the Cisco IP Telephony QoS Design Guide 2 p 1 3 KW Avaya IP Telephony Implementation Guide 24 WAN L3 Packet Size Link Speed 64 bytes 128 bytes 256 bytes 512 bytes 1024 bytes 1500 bytes 56 kbps 64 kbps Table 5 Cisco seralization delay matrix Consult Cisco s documentation for detailed information regarding traffic shaping and LFI and be especially careful with LFI On one hand it reduces the serialization delay but on the other it increases the amount of L2 overhead This is because a single L3 packet that was once transported in a single L2 frame 1s now fragmented and transported in multiple L2 frames Configure the fragment size to be as large as possible while still allowing for acceptable voice quality Instead of implementing LFI some choose to simply lower the MTU size to reduce serialization delay Two possible reasons for this are that LFI may not be supported on a given interface or that lowering the MTU 1s easier to configure As explained in section 2 2 under the heading L3 Fragment
161. turns to VLAN 10 directly without passing through VLAN 1 In this example the VLANTEST 0 option is invoked to make the phone permanently remain on the voice VLAN See section 4 2 heading IP Phone and Attached PC on Different VLANs for a full explanation of how the phone operates between the data and voice VLANs including the use of the VLANTEST parameter The LZQVLAN parameter should not be added to the VLAN 10 DHCP scope This is so that in the event a phone is connected to a port that has VLAN 10 as the port native VLAN it will not receive instructions from the DHCP scope to enable tagging In such a case the phone would not require tagging to function on VLAN 10 and tagging could result in an incompatibility with the Eth switch PC configuration The PC can be statically addressed with a VLAN 1 address or it can receive a VLAN 1 address via DHCP No special configurations are required KW Avaya IP Telephony Implementation Guide 61 Appendix B Cisco Auto Discovery This appendix describes Cisco s proprietary auto discovery feature using CDP and auxiliaryvlan or voice vlan and how they relate to Avaya IP phones Substantial testing and production operation have shown that Avaya IP phones interoperate with both auxiliaryvlan CatOS and voice vlan IOS and these have become the preferred methods of implementation over explicit 802 1Q trunking This interoperability research was initiated because of the inability to enable portfast
162. ty matrix beginning on page 3 of this form KW Avaya IP Telephony Implementation Guide 39 There are network address translation NAT options for direct IP IP audio Since Avaya Communication Manager 1 3 Avaya has permitted shuffling between endpoints that are separated by NAT NAT has been a hurdle for VoIP due to the fact that the address in the IP header is translated but embedded IP addresses in the H 323 messages are not translated This hurdle has been overcome to some extent with the NAT shuffling feature in Communication Manager without the need for H 323 aware NAT devices See NAT Tutorial and Avaya Communication Manager 1 3 NAT Shuffling Feature at www avaya com Note In addition to the ip network region form shuffling and hairpinning must be enabled on two other forms the system parameters features form page 16 and the station form page 2 for each station The RTCP monitoring feature is used with the Avaya VoIP Monitoring Manager VMM Enabling this feature causes the audio endpoints in this region to send periodic RTCP reports to VMM VMM uses these reports to keep a history of audio quality for all reporting endpoints The default server parameters are configured on the system parameters ip options form If the default settings are not desired in any given network region specific settings can be applied on a per region basis The RSVP feature requires careful integration with the IP network and must
163. ue starts to get full randomly discard packets in this random detect dscp based queue based on DSCP lower values get discarded first interface Serial0 description T1 ip address 172 16 0 1 apply the voipQoS policy outbound on this interface Service policy output voipQoS For applications where it is feasible using a single queue for audio and signaling simplifies configuration and reduces router resource consumption operating a single queue consumes less router resources than operating two queues Separate Queues vs Single Queue From a theoretical standpoint using separate queues is ideal When considering the three detriments to IP telephony delay jitter and loss audio is more sensitive to delay and jitter whereas signaling is more sensitive to loss This is not to say that audio is not sensitive to loss or that signaling is not sensitive to delay and jitter but there are fine tuning points that apply to queuing to optimize it for audio or signaling From a practical standpoint in terms of user experience these fine points may matter in some cases and not in others If the amount of signaling is negligible compared to audio and if the size of the WAN link is such that serialization delay is not a factor typically 768k or greater then it is feasible to put audio and signaling in the same priority queue as long as the queue is large enough to sustain both In this case the larger signaling packets do not disrupt audio
164. used by the DEFINITY servers to control the port networks cabinets and port boards Avaya IP Telephony Implementation Guide IP enabled DEFINITY System Adjunct Location Medium Large Enterprise Main Location Ye Ye H 225 RAaAS amp F IP IP Q 931 signaling H 225 KE EPN PPN EPN ull ee CCMS from processor TS L _ to port boards across 4 gt d backplane RTP A C LAN j 5 5 ndi J IP Net ee E C LAN a MedPro aai kas Pi MedPro A iets PEP EPN l Enterprise T Pa TDM bus TDM bus 4T IP Network DCP MCC SCC ST MCC DCP j Anal amp CCMS and bearer Center Stage amp Anal mae over TDM or ATM gt or ATM PNC nang Figure 2 IP enabled DEFINITY System P enabled DEFINITY System is the same architecture as before but with IP port boards added The Control LAN C LAN board is the call servers interface into the IP network for call signaling H 225 which is a component of H 323 is the protocol used for call signaling H 225 itself has two components RAS for endpoint registration and Q 931 for call signaling The IP Media Processor MedPro board is the IP termination point for audio As of Communication Manager 3 0 there is a higher capacity version of the MedPro board called IP Media Resource 320
165. wever Cisco recommends caution in using RTP header compression because it can significantly tax the processor if the compression is done in software Depending on the processor load before compression enabling RTP header compression could significantly slow down or crash the router For best results use a hardware IOS interface module combination that permits the compression to be done in hardware 3 QC 333 5 RTP Header Compression and QoS RTP header compression has to function with exactness or it will disrupt audio If for any reason the compression at one end of the WAN link and decompression at the other end do not function properly the result could be intermittent loss of audio or one way audio This has been very difficult to quantify but there is some anecdotal evidence One production site in particular experienced intermittent one way audio whose cause was very difficult to troubleshoot and isolate When RTP header compression was disabled simply for experimentation purposes the audio problems went away The Test This section details the results of a simple RTP header compression test conducted in a lab environment Although this test was conducted using Cisco routers the expected behavior is the same for any router that performs this function as specified in RFC 2508 7 This test was performed in the following lab configuration KW Avaya IP Telephony Implementation Guide 65
166. y equivalent to QSIG which also rides on top of Q 931 as illustrated between the IP Connect and DEFINITY server Gatekeepers such as the S8700 S8300 and S8500 and Cisco Call Manager in this illustration can connect to one another using IP trunks The medium is IP and the signaling protocol 1s H 323 but Q 931 is still the fundamental component of H 323 that does the call signaling And as with ISDN PRI trunks QSIG or DCS can be overlaid on top of Q 931 QSIG is the standard signaling protocol that provides the feature richness expected in enterprises Generally speaking traditional telephony systems support a broad range of QSIG features while pure IP telephony systems support a very limited range Due to the history and leadership of Avaya in traditional telephony all Avaya call servers whether traditional P enabled or pure IP support virtually the same broad range of QSIG features KW Avaya IP Telephony Implementation Guide 14 1 3 VolP Protocols and Ports The following figure illustrates the protocol stacks relevant to VoIP The contents of the upper layer protocol messages are important to those who develop VoIP applications However those who implement these applications are typically not concerned with decoding the upper layer messages Instead they are concerned primarily with the transport mechanism TCP and UDP ports so that they can verify and filter these message exchanges H 245 CODEC negotiat
167. y established call signaling session TCP 1720 gatekeeper port variable phone port There are some call signaling exchanges on this TCP session Called phone also establishes an audio stream with an audio resource as directed by the gatekeeper but this stream is one way until the call completes Called phone answers resulting in more call signaling activity and the call completes The call could remain in this state but In most cases unless configured otherwise the gatekeeper contacts both phones and instructs them to direct their audio streams to each other Phones direct audio streams to each other as instructed by the gatekeeper One of the phones hangs up resulting in more call signaling activity Gatekeeper contacts both phones signals that the call has ended and instructs them to tear down audio streams Phones tear down audio streams Keepalive Mechanisms There are two types of keepalive mechanisms RAS and TCP RAS keepalive The IP telephone sends RAS keepalive messages to the gatekeeper at a time to live TTL interval specified by the gatekeeper On a protocol analyzer a RAS keepalive message shows up as a RAS Registration Request RRQ with the keepalive bit set in the RAS decode Each request message is acknowledged by the gatekeeper with a RAS Registration Confirm RCF This exchange takes place over the RAS socket which has UDP port 1719 on the gatekeeper side KW Avaya IP Te

Avaya IP Telephony User's Manual

Contents

Download Pdf Manuals

Related Search

Related Contents