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1. STUN SERVER DIF IF COATE PIELICA ng NAT ROUTER mg OR F b p PUBLICA NAT S INTERNET OIE IP FRADA DIR IF PRIVADA DIF IF DIR IF PRIS D FRIYADA TX Ra Registration ina STUN server 84 AEQ PHOENI X ALI O L STUN SERVER DIF IF PUBLICS wire ROUTER PUBLICA oy NAT ROUTER DIR IP ka FLELICA NAT DIR IF DR P PRIVADA PRI s A DIR P ne US PR Td Dic PRIVY ADA Ses SSES TX us Notification from the server of the corresponding NATs detected The response further enables the STUN client to determine the type of NAT being used since different NAT types handle incoming UDP packets in different ways STUN supports three of the four main existing types of NAT Full Cone Restricted Cone and Port Restricted Cone It does not however support Symmetric NAT also known as bidirectional NAT although Phoenix ALIO allows it to be detected and reports its presence to the user Once the client has discovered its public address it can advise its peers of that address STUN is useful as a complement to protocols like SIP SIP uses UDP packets to signal sound video and text traffic over the Internet but does not enable communication to be established when the devices at the ends of the communication circuit are behind NAT routings This is why STUN is customarily used in these applications to permit communication to be established The co
2. AEQ PHOENI X ALI O 88
3. 7 RTP PUBLIC IP parameter that will tell the unit which public IP will correspond to the RTP of its IP interface so that it can send the said IP in its SIP messages The router or firewall administrator must tell you the value of this parameter Usually the administrator will take out SIP traffic and RTP using the same public IP configure in point number 3 For instance 212 170 180 177 8 RTP PUBLIC PORT parameter that will tell the Phoenix ALIO which public port will correspond to the RTP of its IP interface so that it can send the said port in its SIP messages The router or firewall administrator must tell you the value of this parameter in order to make the required port forwarding For instance 8002 In the aforementioned note the detailed configuration and need of port forwarding is explained 4 3 3 AUTO 1 local network audio This mode will be used primarily when two units that are in the same local network need to communicate with each other when the Proxy SIP is on the Internet and it s the one provided by AEQ sip aeq es 4 3 4 AUTO 2 local network audio This mode will be used primarily when two units that are in the same local network need to communicate with each other when a Proxy SIP on the Internet is used it s not the one provided by AEQ sip aeq es and only if the AUTO1 mode doesn t work properly 4 3 5 AUTO 3 audio over internet This mode will be used mainly when you wish to put two units in comm
4. 20 AEQ PHOENIX ALIO 3 7 Function keys The units operational keys are completed by 6 illuminated function keys located at the right area of ALIO front panel Some of these keys IP SIP AUTO CODEC configure parameters or operating modes for the currently active channel only HELP and MENU keys are associated to functions that are common for the unit The detailed description of the function for each key is described below 3 7 1 MODE keys IP and SIP These keys are labeled as IP and SIP and their operation are exclusive only one of these can be enabled at any given time These keys affect the currently selected channel only PROG or COORD and provided that the latter is available licensed They allow the user to select how Phoenix ALIO communicates with other equipment IP mode RTP SmartRTP When the IP key is pressed the unit will establish all communications in RTP mode without using SIP In order to establish a communication in this mode the destination IP address must be known as well as the receiving audio port The standard operation compatible with other manufacturers requires that both sides of the communication initiate the call call the IP and the active audio port of the remote equipment and that both have selected the same encoding algorithm for the communication However thanks to the SmartRTP functionality from AEQ the task o
5. AAC LC o Mode Mono stereo MS stereo Sampling frequency 24 32 48 KHz o Bitrate 32 64 96 128 192 256 Kbps AAC LD o Mode Mono stereo MS stereo Sampling frequency 48 KHz o Bitrate 32 64 96 128 192 256 Kbps Please contact AEQ Sales Department or authorized dealers for more information AEQ PHOENI X ALI O 1 4 Block diagram 1 4 1 Internal diagram The unit s internal design is organized in several functional modules that make Phoenix ALIO audiocodec a complete IP communications platform The unit is provided with a professional quality analog audio input and output system and a versatile audio matrix with processing capabilities A simplified view of the distribution of these modules into the equipment s motherboard is presented below as well as a brief description of each modules functionality Ta d x POWER d LOUT L LIN R AUDIO AUDIO MATRIX PROCESSOR amp MIXER LIN L FPGA HP HP ARM CPL FIXED POINT DSP FRONT PANEL Internal Phoenix ALIO layout detail The audio part ANALOG I O integrates 4 microphone preamplifiers featuring programmable digital gain as well as the line level balanced inputs and outputs together with two stereo headphone outputs able to drive both high and low impedance sets The power supply section POWER converts the DC input 12V 1A to the different DC voltages required inside the unit The FRONT PANEL is c
6. MIC4 and LINE IN inputs are mutually exclusive what means that once the bus send assignment is changed from one of them the other will become inactive The right area of the Mixer window allows control of the three outputs 2 headphones LINE out in a similar way the source can be selected for each output from CUE PROG or COORD buses These buttons are mutually exclusive and when activated they will be filled in the color assigned to the corresponding bus red green or blue respectively as well as in the output amplifier symbols in the General view allowing for a quick identification of the outputs outing at a glance Each outputs volume can be varied between O dB and mute The balance between transmission and reception that the ALIO operator has adjusted on the front panel can be viewed as well but only for informative purposes this adjustment cannot be remotely controlled 4 1 3 Vumeters window Last you can access to the equipment s remote vumeters by clicking on the VU button located below MIX one They will appear into a floating window alio ssm that you can move to the desired position It is 192 169 1 68 possible to open several vumeter windows this is useful for example to check the correct audio transmission between two units that are connected and controlled by the application The maximum PRG number of vumeters that can be simultaneously displayed can be altered in the ControIPHOENIX INI L R
7. SI PRG Ok OK SE hot connected AUTO 92 16 URI 1 alio1S_ prog URI 2 alio15_coor ae 6 5 3 1 6 5 3 2 If the IP a Call SA It is mandatory that the called unit URI is specified in any of the following formats adequate for Direct SIP communications lt equipment s_ name gt lt equipment s_IP_ address gt for example ssanchez 192 168 1 83 If the SIP port of the other end is not 5060 standard SIP port the identifier must include the port in use For example ssanchez 192 168 1 83 5061 VERY IMPORTANT lt equipment s_name gt must not be longer than 19 characters Press the green Call button on the screen to make the call Answer the call on the other unit if Autoanswer option is not active You can observe the changes of status from OK to CONNECTED and synchronized when calling from PROGRAM PUDENIS Lk ALi CONNECTED ssanchezm192 165 1 58 not connected OK not connected Al IO URI 1 alio15_prog URI 2 alio15_coor LE Verify that the SYNC LED beneath the CALL button is lighted in green to indicate that the communication has been successfully established Send audio from one equipment to another verifying that the Tx and Rx ebe audio presence indicators change to green GE amo Ending an IP call in DIRECT SIP mode In order to hang call just click on CALL button of the connected channel in one of both codecs A
8. Functional areas or sections of the user interface are described next CONTROL SURFACE The control surface has been designed bearing Gr OLED displa _ mind that quite often it is not possible to send a p ay qualified technician accomplish an outside Navigation Channel broadcast It can be totally controlled from the station _and also locally operated by choosing among several simple options that are further simplified when connected to another AEQ Phoenix audiocodec a more detailed confi juration can be carried out Y Alphanumeric keyboard KA ef and call buttons a Input controls Mic and lin level adjustments with ONT channel buttons Output controls listening level for headphones 1 and 2 and line output with TX RX balance control 6 FUNCTION KEYS IP RTP Smart RTP modes SIP N ACIP compatible mode AUTO SmartRTP auto answer etc CODEC codecs list HELP remote support MENU advanced options O O 15 AEQ PHOENI X ALI O 3 1 OLED Screen a High contrast OLED screen with wide viewing angle 256 x 64 pixels resolution in gray scale When Phoenix ALIO is powered up after a delay of approx 15 seconds that doesn t indicate any malfunction in the unit two welcome screens will appear sequentially the first one shows AEQ logo and the second one displays the audiocodec model PHOENIX YALIO Detail of welcome screens
9. OK CONNECTED CONNECTING DISCONNECTING CALLING NO LINK REGISTER ERROR REGISTERING CALL ERROR phxCh1TxAudio Ch1 s audio input indicator status Indicates the status of the virtual LED indicating audio presence according to the corresponding configured parameters phxCh1RxAudio Chile audio output indicator status Indicates the status of the virtual LED indicating audio presence according to the corresponding configured parameters phxCh2TxAudio Ch2 s audio input indicator status Indicates the status of the virtual LED indicating audio presence according to the corresponding configured parameters phxCh2RxAudio Ch2 s audio output indicator status Indicates the status of the virtual LED indicating audio presence according to the corresponding configured parameters phxCh1OnAir Ch1 s ON AIR option activated or not phxCh2OnAir Ch2 s ON AIR option activated or not phxCh1Synced Ch1 s audio synchronized or not phxCh2Synced Ch2 s audio synchronized or not phxCh1BackupInterfaceActive Whether or not Ch1 s backup interface is being used on a Call phxCh2BackuplInterfaceActive Whether or not Ch2 s backup interface is being used on a Call mib2 system sysUpTime sysContact sysDescr sysServices Standard SNMP commands indicating things such as equipment s turn on time etc For more information please check MIB II specification in RFC1213 http tools ietf org html rfc1213 For more inf
10. 2 2 3 1 Using an external powerbank as an UPS ccccceeceeeeeeeeeeeeeeeeeeaeeeeeens 13 3 USER INTERFACE DESCRIPTION MANUAL CONTROL cccccessssssseeeeeeesseeeeeesseees 15 Pe OLED ye E 16 3 2 Navigation Channel encoder NAVI CHL 17 E De EE 17 3 4 Alphanumeric keyboard and Call buttons cc ceccceeeeeceeeeceeeeeseeeeseeeeeseeeeseeeeseeeensaees 18 SEET 19 OU US ONE ee EE 20 3 1 le elei 21 3 7 1 MODE keys IP and GI 21 a eae EE e al lt lt ae ee T eee ee ee ee eer 22 3 7 2 1 SMmartRTP AULOANSWED cccccccceeeeceeceeeceeeeeeeeeseeeceeseeeeeeseeeeesseeeeeees 22 3 7 2 2 Auto FANG UD sacs naicsnnicvaewssatinvwnactvcnnesiaticndess te vauendsticaeesaiioceauandiedesestadeaaeadtvans 22 een EE REENEN ER O16 GODEC EE 22 Slds HELP EE 23 Bid 20s MENU EE 24 O f O11 ETHERNET subme MU eeso e Tea n ie 24 3 7 5 2 COMMUNICATIONS submenu s n aannnnennnnsnnnnnsnnnesrnnnnsnrnrsrnrnsnrresenenene 25 3 7 5 3 MAINTENANCE sUbmMEnU EE 27 4 CONFIGURATION AND OPERATION FROM REMOTE CONTROL SOFTWARE 29 4 1 Individual codec Control WINGOW ccccecccceecceeeeee cece cess eeseeeeseeeeseeeseeeseeeseeeesaueeseeeeaes 29 os E e EI Le EN MEMU DEE 30 4 1 2 Mixer Control WINGOW cccccccccececeeeeceeeeseeeeseeeeae cess ceseeeeseeeseueeseeeeseeesaneesneeaaees 32 4 1 3 Vumeters WINKOW cccccccceccceececeeeeceeeecaeeceaeccaeeceuceseueesaueeseeseuees
11. Provide information relating to certain equipment s events that can be considered as alarms They usually have two possible states Active or Inactive The list of alarms defined for Phoenix ALIO is as follows phxCh1NoTxAudioAlarm Audio detection event at Channel 1 s input phxCh1NoRxAudioAlarm Audio detection event at Channel 1 s output phxCh2NoTxAudioAlarm Audio detection event at Channel 2 s input phxCh2NoRxAudioAlarm Audio detection event at Channel 2 s output phxCh1NoAudioSyncAlarm Sync event at Channel 1 while connected phxCh2NoAudioSyncAlarm Sync event at Channel 2 while connected phxCh1CallEndAlarm Call ended on Channel 1 due to incoming RTP traffic loss phxCh2CallEndAlarm Call ended on Channel 2 due to incoming RTP traffic loss phxOtherAlarm Other alarms see name gt Fail to register in SIP PROXY server activation deactivation of BACKUP interface for Ch1 and or Ch2 coldStart alarm Starting from unit off This is a standard SNMP alarm it appears only one time and has no activation or deactivation These alarms are sent whenever they change but we can choose from the SNMP client which ones are shown treated and which not The first 4 audio alarms are configurable and they are activated whenever the incoming or outgoing depending on the particular alarm audio level is below a certain threshold for a given time parameters that are set either by means of the Control Phoenix software
12. REGISTERED or interface status OK CONNECTED CONNECTED_NO_DATA NO_SYNC e CONNECTED TO calling called equipment s name or number identifier number or Unknown when ID is not provided or not connected when there is no established communication The lower part of the individual codec window identifies each unit by its given name IP address and each channel URIs identifier for SIP calls for each of both channels ALIOL WRI 1 alio 15 prog WRI 2 alio15 coor On the right part for both channels PROGRAM and COORDINATION we can find the CALL button indicator and the SYNC indicator as well as two audio presence indicators for both directions transmission Tx and reception Rx 4 1 1 CONFIG Menu At the right side CONFIG button gives access to a configuration menu with the following options General Contacts Ethernet Miscellaneous and Network Just click on CONFIG button again in order to close this menu Configuration k d Ethernet d d AT ZS OD Miscellaneous Network ekra ennon G u 30 AEQ PHOENI X ALI O SA The General option is the most important of the ones associated to CONFIG button you can configure the audio routing and levels from to the equipment the selected audio encoding algorithm the interface to be used from INTERFACE drop down menu and access to Advanced channel configura
13. disconnected 5 3 3 2 Receiving and accepting DIRECT SIP calls If the unit interface is correctly configured and the Autoanswer mode is not active AUTO gt AUTOANSWER OFF when a call is received As opposed to SmartRTP mode incoming SIP calls ARE signaled and unless the Autoanswer option is enabled the user can decide whether to accept or reject the call by means of the OR ESC DEL keys The unit will emit an acoustic signal It can however be disabled under the MENU gt MAINTENANCE gt BUZZER menu The OK key will simultaneously blink to warn the user Information about the caller will appear in the OLED screen indicating the channel PROGRAM or COORDINATION where the call is coming to The user can accept the call by pressing the OR key or reject it by pressing the ESC DEL key assuming that Autoanswer option is not enabled If the call is accepted call status is displayed o CONNECTING The ORT key will blink during this time o CONNECTED NO SYNC NO DATA When the call has been successfully established data is received but synchronization to it is not possible or no data at all is received respectively The OK key will remain steadily illuminated 54 AEQ PHOENI X ALI O If status is NO SYNC or NO DATA and auto hang up option is enabled the call will be rejected after the defined time and the OKT key illumination will turn off Once connected with the remote end verify that
14. 12 keys that among many other functions allows the user to dial IP addresses and port numbers in when calling in RTP mode or to type letters and symbols when in SIP Session Initiation a Protocol mode OK It can also be used to type text just press each key repeatedly to switch between the different letters available for the same key just like you a if would do to type an SMS on a mobile phone ESC DEL Depending on the status the OK key with a green telephone allows the user to initiate a call or accept an option within a menu On the other hand the ESC DEL key with a red telephone can hang up a communication delete or go back in a menu When the OK key is pressed from the idle screen to make a call the alphanumeric keyboard gets illuminated in red when the call is to be made in PROG or in green when the current channel is COORD Dialing in PROG Dialing in COORD The keys are illuminated laterally this way it is possible to determine whether the channel is PROG or COORD even by color blind people NOTE The key shifts between capitals lower case and numbers When you are typing keys 1 and allows you to enter special characters and among them and key 0 can generate number 0 or spaces The key is also a shortcut to switch between MIC4 and LINE provided that we are not typing into a text menu AEQ PHOENI X ALI O 3 5 Inputs control ri Each input has an
15. 3 DIRECT SIP This type of connection is selected when you have a connection with SIP protocol in the signaling phase prior to connection but without the presence of an external SIP server It is necessary to know the IP address of the equipment you want to call in advance but not necessarily the audio ports In order to call in Direct SIP mode you must take into account that for the URI or SIP identifier of the equipment the right syntax is lt unit_name gt lt unit_IP_address gt type for instance ohxalio_231 172 26 5 57 When the correspondent SIP port is not the 5060 SIP Standard port the identifier must include the used port For instance phxalio_ 231 172 25 32 11 5061 When you create a Call Book these fields describing a contact can be modified in the Call Book that can be accessed from a codec individual control window through the Contacts option in Configuration see section 5 1 7 of AEQ ControlPHOENIX user s manual In order to call a same contact using different IP modes as defined in INTERFACE drop down menu different contact entries must be created enra a Ethernet Network You can access the IP configuration submenu for DIRECT SIP mode by clicking on I F Setup button and that it is explained in section 6 1 4 2 of AEQ ControlPHOENIX user s manual e In SIP Parameters submenu you can find User Name enables you to edit th
16. 65 534 Default values 5004 PROG and 5008 COORD o Adaptive Fixed and Adaptive buffer max Fixed buffer length this option allows you to configure the type and maximum size of reception buffer See section 4 4 o Symmetric RTP this option allows you to force the local unit to send audio to the same IP and port from which it is receiving audio The destination port specified when making the call will be ignored when we receive packets from the remote equipment This option will allow you to connect to an audiocodec with unknown IP and or port because it s behind a router with NAT for instance Each unit will send audio to the Local media port of the remote equipment automatically thanks to the SIP signaling protocol That signaling also accomplishes coding profile negotiation and call establishment release from any of both sides of the communication once the remote equipment has been identified by its IP address and reached 4 2 4 Sending audio to multiple destinations Broadcast Multicast and Multi unicast It is possible to send the same audio RTP stream to several different destinations in RTP raw mode see section 4 2 1 There are several possibilities to do so see AEQ ControlPHOENIX manual a Broadcast the audio stream can be sent to all the equipments within a local network only by specifying a special address in the destination address field This address is calculated as the network address with the
17. 9 to 18 V DC Power consumption 12W max External adapter universal 90 263V input Optional UPS with 12V output for the audiocodec and two USB ports for router supply and or mobile devices charging AEQ offers the SmartRTP call initiation protocol in order to greatly simplify the operation of the audiocodec AEQ also offers two SIP servers free of charge as a standard service for the users of Phoenix ALIO One of them is configured as main and the other is provided as a backup More information can be found in Appendix B 1 3 Available encoding algorithms OPUS with Fs 48 KHz mono stereo with 4 mono and 3 stereo presets Bit rates between 12 and 192 Kbps very low delay and audio bandwidth between 6 and 20 kHz G711 A law u law 64 Kbps low delay 3 5 KHz audio bandwidth G722 64 Kbps low delay 7 KHz audio bandwidth AEQ LD with Fs 16 32 or 48 KHz mono or stereo Available bit rates between 64 and 384 Kbps audio bandwidth between 7 and 19 KHz MPEG 1 and 2 LII with Fs between 16 and 48 KHz mono stereo dual channel and Joint stereo Binary bit rate between 64 and 384 Kbps Audio bandwidth between 10 5 and 16 5 KHz PCM linear very low delay transparent quality Fs 48 or 32 KHz with 12 16 20 or 24 bits sample mono or stereo bit rates between 576 and 2304 Kbps audio bandwidth between 16 and 20 KHz Additional encoding modes can be considered according to each customer s specific needs such as
18. After a while the display changes to show the MAIN STATUS screen where the configuration of the different inputs and outputs is presented Ra ZE T SA i p a baal P Hic HIC MHICd x Detail of the MAIN STATUS screen This screen is divided in 7 columns From left to right the first 4 correspond to the 4 active inputs The first three ones are always MIC1 MIC2 and MIC3 The fourth can correspond to MIC4 or LINE IN This can be selected through the MIC4 LINE menu or through the shortcut key in the numeric keyboard the label under the level bar will change accordingly between MIC4 and LINE Finally the three last columns correspond to the outputs HP1 HP2 and OUT as indicated under each level bar Each bar represents a relative mix level or output volume Each MIC input can also display a PH legend above the level bar indicating that the input has its corresponding 12V phantom power supply activated Also the input names can appear highlighted as MIC1 in the above example screen This indicates that a process equalization is applied to the input signal Above each output level bar there is a label showing PGM COOR or CUE These are indicating what program bus is being monitored on each of the outputs This display can also display the different operation and configuration menus that are accessed and browsed through the navigation NAVI Ch rotary e
19. Authentication Passwords 8 digits alphanumeric passwords This configuration is the right one for working with any of both AEQ s SIP servers 83 AEQ PHOENI X ALI O B5 STUN protocol STUN Simple Transversal of UDP over NATs is a network protocol of the client server type that allows NAT clients to find their public IP address the type of NAT where it is located and the Internet port associated with the local port through NAT This information is used to configure a UDP communication between two hosts located behind NAT routers NAT Network Address Translation is a mechanism used by IP routers to exchange packets between two networks that assign each other incompatible addresses It consists of converting in real time the addresses used in the transported packets It is also necessary to edit the packets to enable the operation of protocols that include address information within the protocol conversation It is most commonly utilized to enable the use of private addresses and still provide connectivity with the rest of the Internet PHOENIX ALIO includes a STUN client that sends a request to a STUN server The STUN server then informs the client of its public IP and which port has been opened by NAT to permit incoming traffic to enter the client s network This information enables the Phoenix ALIO to identify its position within the SIP server This protocol is used in AUTO3 and AUTO4 NAT TRAVERSAL modes see section 4 3
20. Male connector Balanced connection i Connector as seen by the soldered 1 2 2 side L output Male R ouput Male XLR 3 pinout Pin 1 gt Ground Pin 2 gt Output Pin 3 gt Output 2 2 Description of the back panel and connections i ip a A TED pisn Pisni pus 4 Vi A N j X j Naa VW Nee bie 2 2 1 Microphone inputs MIC1 MIC2 MIC3 and MIC4 a XLR 3 female connector Balanced connection o Connector as seen by the soldered side 1 Input Female Male plug cable XLR 3 pinout Pin 1 gt Ground Pin 2 gt Input Pin 3 gt Input AEQ PHOENI X ALI O All microphone inputs MIC1 MIC2 MIC3 and MIC4 feature low noise preamplifiers and are able to provide Phantom supply 12 V DC 10 mA These can be enabled from each input s menu see section 3 5 in order to offer compatibility with both dynamic or condenser microphones The range of the preamplifier gain is wide range 0 to 65dB making it suitable for a large range of microphone models available on the market 2 2 2 Ethernet port PHOENIX ALIO features one Ethernet port Using this port the unit can be connected to a LAN or WAN network and send receive audio over IP This port is also used to configure and administrate the unit from one or more computers using the remote control software Please refer to AEQ ControlIPHOENIX application manual The connector is a standard RJ45 10
21. P A ALIO Call on PRG Call to By clicking here the last completed calls S192 168 1 68 5008 am Tri Calls the contacts stored in the call book Contacts or the reachable IP units IP but only those configured in compatible communication modes are listed Channel interface RTP raw Calls LL Equipment Contacts EEN iP w s Replicas in Contact A 192 168 1 68 5008 Channel 1 gt 15 01 2010 12 14 44 192 168 1 68 5008 Channel 1 15 01 2010 12 12 29 00 58 59 Repeat the audio and mode configuration in the other end As SmartRTP is enabled the other codec will automatically connect as soon as it starts receiving audio traffic so it won t be necessary to set the coding mode dial IP addresses or ports etc nor even accept the incoming call 58 AEQ PHOENI X ALI O e Press the green Call button on the screen to make the call e You can monitor the status of the call on the screen CALLING CONNECTING SYNCHRONIZING CONNECTED NO_DATA NO SYNC e Verify that the SYNC LED beneath the CALL button is lighted in green to indicate that the communication has been successfully established e Once the connection has been established with the remote end confirm the presence of transmitted and received audio by checking the audio presence indicators Tx and Rx NOTE In order to make calls to multiple destinations please consult section 4 2 4 of t
22. a remote unit can work abnormally or become unreachable by the remote control software as a result of a communications error or in the unit itself A method has been developed in order to remotely reboot the audiocodec so normal operation is recovered Inside MAINTENANCE at the bottom of the screen you can find SYSTEM REBOOT section By clicking on Reboot button an information dialog will appear warning that the equipment is being rebooted and it will be disconnect for some seconds after acceptance Resetting system please wait a few seconds before connecting again 71 AEQ PHOENI X ALI O 8 TECHNICAL SPECIFICATIONS Audio input and output NN Commutable Phantom 12V 10mA supply Dynamic range gt 90 dB Electronically balanced Line level Electronically balanced Line level Headphone output 2 stereo jacks for high or low impedance headphones with volume control Audiocharacteristics Distortion lt 0 03 linear loop keen depends on chose encoding algorithm gt 100 dB linear no compression _A D amp D A conversion T Communications interfaces Satellite links An external satellite terminal can be connected to the IP interface see application note NA2 3G 4G modem An external 3G 4G USB modem can be connected to the IP interface by means of an homologated portable router See application note NA5B Menu selectable _ S Backup permanent call etc Menu selectable Coding algorithms OP
23. be selected and A connection profiles will be gd veer defined instead containing Bossa one or more modes This is etappe like that because SIP allows E eer the participants in a Y vore communication to negotiate the coding algorithm so the one to be finally used will be limited to those includes in the selected profile This possibility allows the configuration of the parameters associated to the coding to be used in Audio over IP_networks basing on SIP protocol Proxy SIP and Direct SIP modes This option simplifies the selection of the algorithm to be used in a communication because most of the codecs have several tens of encoding algorithms in order to have the higher compatibility with other equipments When a communication is established using SIP signaling the codec negotiates the use of the first compatible encoding algorithm included in a list called SIP CODEC PROFILE That s why you should put these algorithms in order of preference Each one of the stored entries includes an alphanumeric identifier and a list of algorithms to use organized in order of preference There several preset profiles in the unit grouped by delay quality etc Profiles can be added modified or deleted in the Encoding Profile aye Management SIP screen accessible from the Tools menu in fin SE the upper Menu bar described in paragraph 5 1 8 of AEQ Ser SE ControlPHOENIX user s manual 9 Encoding Profile Managem
24. case as NAOE but adapted to Phoenix Mobile D2 Special applications using different kinds of Internet physical accesses or point to point connections Application note AN1 Connecting a Phoenix Studio and Mobile to Internet through a PC via a WiFi network Application note AN2 Connecting two Phoenix Mobile units using a BGAN satellite link Application note AN3 Connecting two audiocodecs Phx Studio Phx Studio amp Phx Mobile Phx Studio using a private WiMAX network Application note AN4 Connecting two Phoenix Studio units using a dedicated point to point IP radio link Application note AN5 Connecting a Phoenix Mobile to Internet using a 3G router Application note AN5B Connecting Phoenix IP audiocodecs to 3G 4G networks 87 AEQ PHOENI X ALI O APPENDIX E Additional information NOTE This equipment complies with the limits for a Class A digital device pursuant to part 15 of the FCC Rules These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instruction manual may cause harmful interference to radio communications Operation of this equipment in a residential area Is likely to cause harmful interference in which case the user will be required to correct the interference at his own expense
25. confirmation message will appear and the call will be disconnected after acceptance Receiving and accepting IP calls in DIRECT SIP mode interface is correctly configured and automatic answer mode is OFF when you receive The unit and the application will provide acoustic warning This can be disabled for the unit at Configuration gt Miscellaneous gt Buzzer and test The CALL button red LED of the called channel at the individual codec control window in the remote control software corresponding to the unit that is receiving a call will blink at the same time to warn the user In addition if Autoanswer option is not active an incoming call window will appear showing the URI identifier of the caller unit ControlPHOENIX Incoming Calls AEQ To equipment Channel 04 10 2012 08 53 25 sip phoenixMaster sip aeq es internet 172 26 5 59 Channel 1 65 AEQ PHOENI X ALI O The call will be accepted by clicking on the individual codec control window CALL button of the called channel or alternatively on the Accept button in the incoming calls window The screen will show the status of the call o CONNECTING o SYNCHRONIZING o CONNECTED NO_SYNC NO_ DATA Verify that the SYNC LED beneath the CALL button corresponding to that channel is lighted in green to indicate that the communication has been successfully established Send audio from one equipment to another verifyi
26. drive both high and low impedance models with 74 TRS or jack connectors and 1 balanced line output with 2 male XLR connectors All of them are located in the right panel of the unit Each output has a labeled control section HP1 HP2 and LINE OUT located at the bottom right area of the front panel TX RX level balance potentiometer Allows the user to SOURCE continuously adjust which ratio of the corresponding output comes from the channel local transmission TX send and H what from the same channel reception reception RX b When in the central position equal levels of RX and TX 3 will be listened to both with 6dB attenuation Output level encoder Adjusts the listening level of the corresponding output It will be displayed on the OLED with level bars Pressing the output level encoder button cyclically selects the source bus for that output bus between CUE SOURCE LED off PROG SOURCE LED illuminated red or COORD SOURCE LED illuminated green The display will show a label above the corresponding level bar PROG 90 P Lea T e bel as rs CUE COOR In the main screen the display will show the level and source selected for each output Operation is the same for the potentiometers encoders SOURCE LEDS and main display for all HP1 HP2 and LINE OUT outputs NOTE Adjust the volume with caution excessive listening levels can damage your hearing
27. equipment s_name gt lt SIP_realm gt for example phxalio_231 sip aeqg es or phoenixMaster sip aeq es o lt equipment s name gt lt SIP_server_IP gt for example phxalio_231 232 168 1 2 or phoenixMaster 232 168 1 2 where 232 168 1 2 is AEQ SIP Server sip aeq es IP address o lt equipment s_ name gt lt SIP_server gt lt Port gt when SIP port is not 5060 the one used by default in SIP SERVER mode For example ohxalio_231 sip aeqg es 5061 VERY IMPORTANT lt equipment s_name gt must not be longer than 19 characters Press the green Call button on the screen to make the call Answer the call on the other unit if Autoanswer option is not active You can observe the changes of status from OK to CONNECTED and synchronized 61 AEQ PHOENI X ALI O P a 6 5 2 1 6 5 2 2 OK Be not connected pp OK CONNECTED not connected 192 168 1 84 5004 OK Ss not connected C O N F I G 192 168 1 83 Verify that the SYNC LED beneath the CAL button corresponding to that channel is lighted in green to indicate that the communication has been successfully established Send audio from one equipment to another verifying that the Tx and Rx audio presence indicators change to green elo H If the unit is registering in SIP server and the call is being made but no audio comes through please check NAT TRAVERSAL configuration see se
28. equipment part filled with 1 s For instance if the IP address of our codec is 192 168 20 3 and network mask is 255 255 255 0 the corresponding broadcast address is 192 168 20 255 However if the network mask was for example 255 255 0 0 then the broadcast address would be 192 168 255 255 The audio will be sent to a given port so the receiving equipments should have local media port set up to this same port so they are able to receive the RTP stream This mode is not recommended for big networks and is usually blocked by the switches and routers so its use is restricted to small well managed local area networks b Multicast it is also possible to send the audio stream to a special multicast address for example 239 255 20 8 If the receiving equipments call to that same IP they will receive the audio that is being sent provided that their local media port matches the one the transmitter is sending the packets to Phoenix ALIO implements IGMP Internet Group Management Protocol in order to subscribe to multicast group Similarly to broadcast multicast traffic is usually blocked by switches and routers so its use is restricted to local area networks too c Multiple unicast Phoenix units can send the same RTP stream to several different IPs by replication of the encoded audio This can traverse switches and routers in the same way it would do if it was a simple unicast RTP Raw stream although it is limited to a cert
29. has enough charge 13 AEQ PHOENIX ALIO Once all accessories are connected make sure that the powerbank s output voltage is set to 12V and turn it on using its center button If it is deeply discharged please provide charge for some minutes without ALIO connected until the 12V indication remains ON once the button is pressed This device also features additional USB ports One of them remains free even when using AEQ UPS adapter and can be used to charge phones producing a logical reduction in the duration of the battery or to power an external 4G modem router Please check application note AN 5B for more details NOTE It this powerbank is purchased locally please contact AEQ to obtain the special UPS adapter for ALIO CAUTION Due to risk of fire or explosion avoid exposure of the powerbank to shocks temperatures above 45 C liquid pouring etc The unit should be opened ONLY by qualified personnel Please read the manufacturer s recommendations for more details AEQ PHOENI X ALI O 3 USER INTERFACE DESCRIPTION MANUAL CONTROL Configuration and operation of the Phoenix ALIO unit can be done either using the equipment front panel controls featuring an OLED screen and associated controls and indicators or remotely using the AEQ ControlPHOENIX application Control and Configuration software for AEQ Phoenix STRATOS STUDIO MERCURY VENUS VENUS V2 and ALIO audiocodecs This chapter describes the first mode
30. or using the following SET commands For example alarm phxCh1NoTxAudioAlarm will become Active whenever audio from chi input has a level below the threshold defined by phxCh1TxAudioThreshold during a longer time than specified by phxCh1TxAudiolnterval 2 Configurations SET adjustments related to some of the above defined alarms the SNMP client will configure them by means of SET commands although in the case of Phoenix units they can also be modified by means of AEQ ControlPHOENIX remote control software phxCh1TxAudioThreshold Audio threshold for channel 1 s input phxCh1TxAudiolnterval Audio interval for channel 1 s input 47 AEQ PHOENI X ALI O phxCh1RxAudioThreshold Audio threshold for channel 1 s output phxCh1RxAudiolnterval Audio interval for channel 1 s output phxCh2TxAudioThreshold Audio threshold for channel 2 s input phxCh2TxAudiolnterval Audio interval for channel 2 s input phxCh2RxAudioThreshold Audio threshold for channel Ze output phxCh2RxAudiolnterval Audio interval for channel 2 s output 3 Information messages GET showing a status they don t arrive spontaneously or are activated deactivated like the Alarms but they are requested by the SNMP client by means of GET messages phxCh1Status Channel 1 s status gt OK CONNECTED CONNECTING DISCONNECTING CALLING NO LINK REGISTER ERROR REGISTERING CALL ERROR phxCh2Status Channel 2 s status gt
31. represents a group of eight bits translated into decimal form that is whose minimum value is 0 0 0 0 and whose maximum value is 255 255 255 255 IP addresses are classified in two major groups static and dynamic e t is typical for a user to connect to the Internet from his or her home using an IP address This address may change when the user reconnects and this manner of assigning IP addresses is called a dynamic IP address normally abbreviated as dynamic IP e The Internet sites that by nature need to be continuously connected generally have a Static IP address as with the dynamic address a similar abbreviated form is used Static or fixed P that is an address that does not change over time Another possible IP address classification can be made according to address validity e Public IP addresses that are valid in the entire Internet network Currently due to the poor management that has traditionally been applied to the available IP addresses they are a scarce highly costly resource e Private addresses that are only valid in a closed section of the IP network typically corporate and not subject to free access with only one point of connection to the Internet called a gateway constituted by a router B2 2 Unicast vs Multicast Unicast is the transmission of information from a single sender to a single receiver It is distinguished from multicast transmission to certain specific recipients more than one fr
32. s connection e At I F Setup fill in the Local media port where the unit expects to receive RTP audio traffic at If you enable Symmetric RTP mode the unit will send audio to the same port where it is receiving it from This is sometimes useful to overcome NAT routers Local media port 5004 G Adaptive Adaptive buffer max MS 1 Fixed The same screen allows you to configure the type and size of the receiving buffer and FEC parameters as a function of the IP network quality so we have the shortest delay while audio cuts are minimized or eliminated in poor quality networks see paragraph 4 4 of this manual in order to select the optimal buffer configuration depending on your application 63 AEQ PHOENI X ALI O De Return to the general configuration screen check that the omrceopnp selected encoding profile in the green ENCODER area Paton corresponding to that channel PROG or COORD is or correct or otherwise click on Select codec to change it There are several pre defined profiles containing several Bene particular algorithms each one with preference ordering They can be edited and more profiles can be added The called unit will accept the call using the first coding algorithm that it supports from the list independently of the profile selected in that unit at that time Advanced Decide whether you will use the advanced automatic connection options or not ControlPHOENIX dh PRG Advanced conf
33. that enable the user to inform the proxy servers of his or her location For complete information on the SIP protocol we recommend consulting http tools ietf org html rfc3261 B4 1 Working modes With the PROXY SIP option activated in the Phoenix ALIO when the unit is started up it will automatically connect and register itself in the SIP Proxy server configured in its memory indicating its name URI name domain and position IP address To establish any communication the unit that wishes to establish the connection will search the SIP Proxy server register for the information regarding the called device and will redirect the call in a way that is transparent to the user toward the real physical place where the device is located 81 AEQ PHOENIX ALIO SA L SIP SERVER 000 oom 3 1 000 nma e 530 TT de 08 90 SIP SERVER PELLETTI Seele aug Gla seo d A SIP protocol operation diagram Phase 1 Registration Phase 2 Search for the called device in the SIP server database Phase 3 Establishment of the connection AEQ PHOENI X ALI O This working method supported by external SIP servers enables the physical position of a device to be made independent from its logic identifier and through the use of the SIP protocol makes it unnecessary to know more data regarding the called device than its URI During the establishment of the communication phase the encoding algorithm is negotiated simultane
34. to the system Proxy SIP Account enables you to select a Proxy SIP account from a previously created and stored list In case an account is selected the parameters described next would be automatically loaded confirmation is requested Proxy Provider enables you to select the external SIP server with which the unit will work from a previously stored list By default AEQ server will be selected Authentication enables you to edit the password and security information for the user profile associated with the unit in the previously selected SIP server By default the data configured in this field in order to use AEQ server are the following o User the User Name configured in Factory phxalio_231 for instance o Pwd the Password associated to that user o Realm the domain where the SIP Server is located by default Sip aeq es You can find the NAT mode selection at NAT Traversal submenu NAT Traversal is a set of tools used by the equipment in order to surpass the NAT Network Address Translation performed by some routers We can select among several modes depending on the kind of network the unit is connected to Phoenix ALIO offers a total of six different operating modes when traversing devices with NAT routers firewalls Each one of those modes is suitable for a different scenario For instance when the units that are establishing communication are in the same local network the internal working w
35. you were typing a SMS on a mobile phone The URI must be specified according to one of the following formats o lt dest_URI gt lt dest_IP gt Le phoenixMaster 172 26 33 15 o lt dest_URI gt lt dest_IP gt lt SIP_port gt Le phoenixMaster 172 26 33 15 5061 Select the CALL option or press the Ok key again Accept if necessary the call in the other end see 5 3 3 2 The OLED screen displays the call status as well as the destination URI address o CONNECTING The OR key will blink during this time o CONNECTED NO SYNC NO DATA When the call has been successfully established data is received but synchronization to it is not possible or no data at all is received respectively The OK key will remain steadily illuminated If status is NO SYNC or NO DATA and auto hang up option is enabled the call will be rejected after the defined time and the OKT key illumination will turn off Once connected with the remote end verify that the vumeters in Phoenix ALIO front panel show the presence of send and received audio and adjust levels as necessary 5 3 3 1 Ending a DIRECT SIP call In order to finish the communication just press the ESC DEL key for a longer time making sure that the currently selected channel is the one we want to cut The ESC DEL key will blink red during disconnection and the display will show the DISCONNECTING status Both will disappear only when the call has been completely
36. 100 BT type with the following pin assignment Pin 4 Pin 5 BLUE BLUE amp WHITE Pin 3 Pin 6 WHITE amp GREEN GREEN Pin 2 Pin f ORANGE WHITE amp BROWN Fin 1 WHITE amp ORANGE Pin 6 BROWN RJ45 connector pinout 2 2 3 Power supply c PHOENIX ALIO can be powered by an external specifically designed 12V DC power supply The unit can be connected to the provided charger either directly or using an optional UPS in cascade It cannot be connected directly to a vehicle battery without connecting the mentioned optional UPS or an equivalent voltage stabilizer The power cable termination is fitted with a special connector featuring a locking mechanism to prevent accidental disconnections 2 2 3 1 Using an external powerbank as an UPS An external small and portable battery has been homologated as an optional accessory to Phoenix ALIO It can operate as an UPS and also provide a certain degree of portability to the unit as it can provide supply for full operation during about 2 hours when fully charged The recommended model is MP 10000 from XT Power In order to use this powerbank as an UPS a specifically designed adapter must be used This adapter can be purchased from AEQ and allows for charging of the battery at the same time as the PHOENIX ALIO gets power using the same AC DC power adapter supplied with the unit In case that a mains cut happens no operation interruption will be produced as long as the battery
37. 68 1 55 Shee 192 1685 1 55 S64 os PO Pci Hic Mics Mica HFL WFE OUT Pcl Mice Mics Mica HFL HFe Out Piel Mice Mics Mica HFL HFe Gut If status is NO SYNC or NO DATA and auto hang up option is enabled the call will be rejected after the defined time and the OKT key illumination will turn off Once connected with the remote end verify that the vumeters in Phoenix ALIO front panel show the presence of send and received audio and adjust levels as necessary 5 3 1 1 Ending an RTP IP communication e In order to finish the communication just press the ESC DEL key for a longer time making sure that the currently selected channel is the one we want to cut If as recommended SmartRTP is activated in both codecs involved there is no need to repeat the hang up process in the remote end The ESC DEL key will blink red during disconnection and the display will show the DISCONNECTING status Both will disappear only when the call has been completely disconnected CUE CUE CUE DISCOMHECT IHG 192 165 1 55 5664 EN CODEC HEH AD MICL MICE MICS MICd HFL HES out 51 AEQ PHOENI X ALI O 5 3 2 Establishing an IP call in PROXY SIP mode When for instance due to compatibility issues with other manufacturers SIP signaling is required relying on an external server to abstract from involved IP ports etc the PROXY SIP mode will be used Check that the unit is ON Check that the RJ45 cab
38. AEQ PHOENIX ALIO Portable IP Audiocodec that is easy to configure and use Optimized for OPUS encoding algorithms USER S MANUAL ED 10 15 V 1 1 01 12 2015 Firmware Versions CPU 5 20 DSP 3 33 FPGA 5 54 or higher Software Version AEQ ControlIPHOENIX 2 2 0 4 or higher CONTENTS T INTRODUC HON cating rr amen ee ose een acdsee 5 1 1 General ESCO OM EE 5 1 2 Technical CAL ACC ISUCS E 5 1 3 Available encoding algorithMS ccccccceeeceseeeeeeeeeeseeeeeeeeeeeseeeeeseeeeseeeeesneesseeeeseeeeesneeeas 6 RE Block GIG a EE T 1 4 1 Internal diagram KENNEN ENEE T elle T EE 8 1 5 Compatibility with other AEQ codeces 10 1 6 Compatibility with third party codecs cccccccccecseeeeeceeeeeeeeeeecesaeeeeeeeeeeeesaeeseeeseeeeeesaaaees 10 2 PHYSICAL DESCRIPTION OF THE UNIT ccesssecccenseeeccnssesecnseseeenssseeensseseennseeess 11 2 1 Description of the right panel and CONNECTIONS c ccc ceeeccceeeeceeeeeceeececaeeeeseeeeesaeeseaess 11 2 1 1 Headphone 1 and 2 outpoute ccc ceeccceseeceseecseece cece eecaueeaeeseeeeeseeseueesueesaass 11 2 122 LINE mpu LINE IN ON 11 2 1 3 LING outputs CLINE OUT J EE 12 2 2 Description of the back panel and CONNECTIONS ccccceccceeeeecaeeeecaeececeeeeeseeeeseeeesaeees 12 2 2 1 Microphone inputs MIC1 MIC2 MIC3 ang MIC A 12 EE E gege nomnccemasneatestarsace EE E a E A EE sap usaceanetioasd E 13 A PONO SUMO E 13
39. CALL button corresponding to that channel is lighted in green to indicate that the communication has been successfully established Once connected to the remote end verify that the Tx and Rx audio presence indicators change to green 62 AEQ PHOENIX ALIO 6 5 3 Establishing an IP call in DIRECT SIP mode e Ensure that the equipment is powered up and controlled by the software e Establish the appropriate audio configuration mixer e Check that there is incoming audio to the channel PROG or COORD we are going to use to establish the communication the Tx indicator in the individual codec control ay Co window In the general configuration screen and in the list view will change to green e Go to general configuration screen and configure INTERFACE as DIRECT SIP e Enter I F Setup and click on SIP Parameters Check that User Name and Display name are configured User name and IP address constitute the equipment s required connection information ControlPHOENIX Configuration PRG SIP parameters ALIO Local uri Direct SIP A User name phxalio_14 _ displayname Phoenix Channel 1 e Select the working mode to traverse NAT devices NAT Traversal that is more adequate for the network the unit is connected to NOTE It is recommended that you follow Application Notes 0 A or 0 C according to the type of equipment
40. ENI X ALI O Available coding algorithms list in Phoenix ALIO Please contact us to check availability of other algorithms JL 28 IL MOER pr 2 ICE OPUS eo oo a E AEQ LD M ST JST 10 5 16 5 H sma REGELE me IA a OS MPEG 2 Layer PCM 12 16 20 24 bit sample Other different encoding modes can be taken into account according to specific needs of each client 19 AEQ PHOENI X ALI O APPENDIX B Protocols associated with IP communications Communication over IP networks differs notably from the communications traditionally used to date in broadcast environments whether they are POTS or ISDN in that IP networks do not have dedicated resources or qualities of service implemented in most systems with the associated problems this involves in terms of communication signaling establishment maintenance and cleardown This set of problems originates in the technical characteristics that are intrinsic to the definition and operation of communications systems based on IP protocols The EBU TECH 3326 standard developed by the N ACIP working group provides certain tools for attempting to simplify work by making use of many protocols associated with IP communication and which will be described below N ACIP e Signaling understood as connection initiating and ending procedures as well as negotiation of connection parameters encoding algorithms ports etc o SDP Session Description Protocol to describe the para
41. IP signalling packets port depens on the remote unit network not Phoenix network kel keng 60000 Vumeters protocol only when remote control PC is not in Phoenix network Le Remote control protocol only when Phoenix must be connected to PC and PC is not in its network Port depends on PC not on Phoenix Input permissions in router firewall Phoenix unit will have to be able to receive packets from units installed out of the private network Therefore firewall will have to allow that packets sent to Phoenix unit IP to the following ports are received Protocol Port number Usage SNMP protocol monitoring RTP protocol audio packets towards the remote unit may Ca ports depend on Phoenix configuration 5060 5062 SIP protocol SIP signaling packets port depends on M the remote unit network not Phoenix network HTTP protocol only when PC must be connected to Phoenix web server for a firmware upgrading for instance TCP 4422 Remote control protocol only when PC must be connected to Phoenix and PC is not in its network 3 When router firewall uses NAT translation between private and public addresses then a Port Forwarding must be made in the router for each one of the ports described in section 2 and for each one of the IPs of Phoenix units installed in that private network In that case remote unit will send its packets towards router IP its a public IP and the ports configured in router by means of
42. L R Tx Rx controlPHOENIx file located in the application folder for example the following lines specify that the maximum allowable number of vumeters is 10 Vumeters MaxVuToShow 10 If you try to open a higher number of vumeter windows the first one will be closed Click on the right top cross in order to close a vumeters window The represented vumeters correspond for both channels PRG PROGRAM CRD COORDINATION and from left to right to audio transmitted to the channel L and R and received from it also L and R 33 AEQ PHOENI X ALI O 4 2 Connection modes In order to establish an IP communication using PROGRAM or COORD channels first we need to choose one of the three available connection modes PROXY SIP DIRECT SIP and RTP Point to Point RAWY from the INTERFACE drop down menu of the desired channel This is the same as selecting the communications mode with the front panel SIP IP and the use or not of an external Proxy under MENU gt COMMUNICATIONS gt SIP We can access the IP configuration submenu by clicking on I F Setup This menu is described in sections 6 1 4 2 and 6 1 4 3 of AEQ ControlPHOENIX user s manual It is important to know the details of each type of connection so they are explained below 4 2 1 RTP Point to Point RAW This type of connection is selected when the connection over IP will be an RTP type link with calling of the IP address
43. Management of the existing profiles can only be done by the control software and they remain stored in the unit The profiles are stored in the non volatile part of the ALIO s memory From the front panel of the ALIO the user can only select among a list of stored profiles When the user press and hold the CODEC key for more than 2 seconds an informative screen will show up describing the currently selected codec or profile for the active channel without making any modifications to it 3 7 4 HELP key This key that is independent on the active channel is not following the active channel and it sends a notification to the remote control software provided that this is in use This way a Phoenix ALIO user that requires assistance or has doubts can ask the operator that is remotely controlling or monitoring the unit with the AEQ ControlPHOENIX software for help When the HELP key is pressed a notification will appear on the OLED screen of the ALIO and the key will start to flash in red On the remote control software a pop up notification will appear asking the operator to get in touch with the ALIO user 23 AEQ PHOENIX ALIO Help request from ALIO1 Audicodec ALIO1 is asking for your assistance Please contact its operator as soon as possible to provide help Click on OK to acknowledge this warning Once this notification is confirmed by the remote operator the ALIO s HELP key will stop flashing
44. NIX AEQ Phoenix STRATOS STUDIO MERCURY VENUS VENUS V2 and ALIO Audiocodecs Configuration and Control Software The version that is provided together with the equipment 2 2 0 4 or higher can control up to 2 units per software instance If you need to manage more than 2 Phoenix audiocodecs at the same time please contact AEQ sales department to purchase a multicodec license for AEQ ControlPHOENIX Please have the user manual of this application at hand Install and configure it and add the equipment to the controlled equipment list in order to follow the explanations provided in this and following chapters step by step This manual will describe particular ALIO options only as well as some important procedures while the detailed operation is detailed in AEQ ControlPHOENIX user s manual 4 1 Individual codec control window The individual codec control window is thoroughly described in chapter 6 of AEQ ControIPHOENIX user s manual The screen corresponding to Phoenix ALIO is as follows PHOENIX bh ALIO OK Ee not connected CONNECTED 192 168 1 84 5004 ALIOL WRI 1 aio 1S prog URI 2 alio 15 coor Se C a N F I G The name assigned to the device can be seen in the lower area of the window ALIO1 in this example as well as the URI corresponding to both channels URI 1 gt PROGRAM URI 2 gt COORDINATION Additionally a blue link is presented with the equipment s IP address By c
45. PUTS l L R L HP1 Audio matrix structure CGD an L R HP LIN OUT The assignment of mono and stereo signals is performed automatically for example if MIC 2 is routed to the transmit bus of a channel where a stereo coding algorithm has been selected the unit will make a crosspoint to both L and R However if the line input is sent to a mono channel a crosspont creating a L R sum attenuated by 6dB will be accomplished Just in the same way if we want to listen to a mono encoded channel in the headphones the signal will be routed to both sides The send buses are accessed through back lit keys and their colour also denotes to what SEND bus the corresponding audio input routed RED Program GREEN Coordination off CUE The same colour convention is used for the outputs The user can select what bus to monitor by simply pressing on the corresponding rotary encoder The control of the listening level is done with the associated rotary encoder and is also visible through level bars on the OLED display TX RX mixing level is controlled with potentiometers associated to each output AEQ PHOENI X ALI O 1 5 Compatibility with other AEQ codecs PHOENIX ALIO offers the possibility to connect to other AEQ codecs It is compatible with Phoenix MERCURY STUDIO VENUS VENUS V2 STRATOS MOBILE LITE POCKET and PC The SmartRTP mode and OPUS encoding algorithm can be used with Phoenix MERCURY STUDIO VENUS VENUS V2 and STRATOS u
46. Port Forwarding these are public ports not to each Phoenix ports these are private ports 86 AEQ PHOENI X ALI O APPENDIX D Application notes guide This index tries to give users guidance on selecting the most advisable application note in order to connect two audiocodecs of Phoenix family depending on its requirements and working environment Each application note describes the way to configure each of the audiocodecs When both ends are different for instance at one end there s a Phoenix Mobile and at the other end a Phoenix STRATOS different application notes should be followed in order to configure each one All notes are available in electronic format in the CD furnished with the unit D1 Internet connection using standard cable access Application note ANOA Phoenix Studio audiocodec directly connected to Internet by means of a dedicated cable Modem with DHCP SIP call using AEQ SIP Proxy Application note ANOB Same case as NAOA but adapted to Phoenix Mobile Application note ANOC Phoenix Studio Audiocodec connected to a LAN together with other IP equipments connected to Internet by means of a router with NAT that can be configured or we have access to the Network Manager SIP call using AEQ SIP Proxy Application note ANOD Same case as NAOC but adapted to Phoenix Mobile Application note ANOE Same case as NAOC but calling in SIP DIRECT but with no SIP Proxy involved Application note ANOF Same
47. Raw_mode a control will appear in the general configuration window that allows for the activation deactivation of the transmitted stream to the IP channel Make sure that only one of the units connected to the multicast transmitter has this checkbox activated 40 AEQ PHOENIX ALIO Tx DECODER Ty DECODER Coding Coding Select THE MPEG L2 128kbs Select MPEG L2 128Kbs codec LI 45kHz MONO codec L 45KHz MONG es Tx PROG Tx PROG Channel transmission active Channel transmission disabled NOTE 2 Advanced contacts those allowing specification of the communication mode coding algorithm profile replicas etc can only be stored in the General agenda that is saved in a database in the control PC These contacts can however be copied to the different devices but the advanced fields interface coding algorithm SIP account provider and replicas will be lost so only the contact name and contact data main IP port or destination URI will be stored NOTE 3 It is possible to use multiple unicast transmission with SmartRTP active The transmitter unit must be the one that generates the calls and when it hangs up it will send notifications to the MAIN destination only not in the replicas As a consequence only this one will hang up If the user needs that all receivers hang up too the Auto Hang Up option can be activated on them defining a reasonable time lets say 5 10 seconds If on the other
48. S miertace cece ccccssseecceeseeeceeeseecceaeeecsaeecseuseeessaseeessageeesseaes 49 ERR Tele ie Re EE 49 5 3 Ee Ee a Lef Beil e e DEET 49 5 3 1 Establishing an IP communication in RTP mode using SmariRTP c ceeees 50 5 3 1 1 Ending an RTP IP communication 51 5 3 2 Establishing an IP call in PROXY SIP mode cccccecceeceeeeeeeeeeeeesaeeeeeeaeeeeeeaees 52 E CN ER Ending a PROXY SIP TEE 53 5 3 2 2 Receiving and accepting an IP call in PROXY Glbmode 53 5 3 3 Establishing a DIRECT SIP call 0 0 ccccccecceeeeeeeeeeeeeeaeeeeeseeeeeesaeeseesaeeeeeeseees 53 5 3 3 1 Ending a DIRECT RUE 54 5 3 3 2 Receiving and accepting DIRECT SIP calls nnnnannnnaannnnennnnnnnnenenennn 54 6 QUICK START GUIDE REMOTE CONTROL cccctteecceeeeeeeeeeeeeenseeeeeeeeeeeesneeeeeeeenenneneeeees 56 SEN EE CUIDIME MI COMME COINS E 56 6 1 1 Power SUD DIY EN 56 6 1 2 COMMUNICATIONS interface cece c ccc eeeeeeceeeeeecaeeeeeecaeeeesaeeeesseeeeeeseaeeeeeaeeeeeesaass 56 6 2 Turning the unit On EE 56 6 3 Setting up a computer to Control the unit 0 0 cece cecccceeeecceeeeeeeeeesaeeceseeeeseeeeeseeeeesaeees 56 E AUO eee eee E O E N ee ee E E E E ee 57 6 5 Establishing an IP Communication 57 6 5 1 Establishing an IP communication in RTP mode using SmartRTP 000 57 6 5 1 1 Ending an IP communication in RTP mode ccsecccseeeeeeeeeeeeeeeeeeees 59 6 5 2 Establishing an IP
49. US G 711 G 722 MPEG Layer 2 PCM See APPENDIX A Controlanddatainterface S O General characteristics S Display 1x OLED 128x64 pixels 16 grayscale levels 2 x stereo vumeters 20 multi color LEDs Characteristics are subject to changes without previous notice Safety regulations CE Marking Electromagnetic compatibility according to EU directives EN 50081 1 EN 50052 2 72 AEQ PHOENI X ALI O 9 A E Q WARRANTY AEQ warrants that this product has been designed and manufactured under a certified Quality Assurance System AEQ therefore warrants that the necessary test protocols to assure the proper operation and the specified technical characteristics of the product have been followed and accomplished This includes that the general protocols for design and production and the particular ones for this product are conveniently documented 1 The present guarantee does not exclude or limit in any way any legally recognized right of the client 2 The period of guarantee is defined to be twelve natural months starting from the date of purchase of the product by the first client To be able to apply to the established in this guarantee it is compulsory condition to inform the authorized distributor or to its effect an AEQ Sales office or the Technical Service of AEQ within thirty days of the appearance of the defect and within the period of guarantee as well as to facilitate a copy of the purchase invoice and ser
50. a window where all parameters of the ALIO mixer can be controlled level and gain for all inputs input routing tone adjustments MIC4 or LINE IN selection output routing and level etc Phoenix ALIO mixer OUTPUTS AEQ PHOENI X ALI O Audio can be controlled in real time from this screen alternatively or in parallel with the unit front panel from the control PC that can be located either at the side of the unit or remotely MIC1 2 and 3 input channels feature the following controls Send assignments allow the user to select which bus the corresponding input is sent to PROGRAM COORDINATION or CUE These buttons are mutually exclusive and when activated they will be filled in the color assigned to the corresponding bus red green or blue respectively as well as the input amplifier symbols in the General view allowing for a quick identification of the inputs routing at a glance Mix fader allows adjustment of the corresponding input mix to the selected bus between mute and 18dB The selected value is shown below the fader at the right of PH button Tone controls bass and treble permitting an adjustment between 12 and 12dB individually for each input PH Phantom button activates or deactivates Phantom supply 12V 10mA for the corresponding microphone Gain clicking this button opens a window to control the gain of each preamplifier in a range from 0 to 65dB for microphone inputs only
51. access Wi Fi 3G 4G and satellite networks as described in specific application notes available on AEQ s website aeq eu or aeqbroadcast com 1 4 2 Audio matrix The different audio sources are first converted to digital format 24 bits 48 KHz sampling rate to be processed by a digital audio matrix Once the outputs are obtained they are converted back to analog format FPGA Coprocesador Ge Routing mixing Loi HG AO Audio input output structure The following figure shows the internal buses for transmission reception and CUE of both communications channels that are calculated in the audio matrix so the user can better understand what can be done with the different audio sources and what can be listened at the different outputs As can be seen each input 4 microphones or 3 microphones LINE IN can be routed to any of the transmit buses program or coordination or to CUE pre listen bus At the same time we can select what to listen at any of the outputs LINE OUT 2 stereo headphones Program or Coordination buses choosing the mix level between send and return directions or the local CUE pre listen bus AEQ PHOENI X ALI O MIC1 MIC2 MIC3 MIC4 LIN L LIN R inputs IC ene DRS Audio inputs processor Matriz de audio L TX PROG PROG channel k L RX PROG WU interface L K DR IO E E SE TX COORD R oma TTT L Ze NES a ER E ee RX COORD EERENS l EETEEENY a Matriz de audio D A 6ch OUT
52. address and port of the remote unit SIP mode will alternatively change between DIRECT SIP and PROXY SIP when pressing the SIP key repeatedly Depending on the selected mode the communications menu options MENUSCOMMUNICATIONS SIP may vary slightly so the user only needs to configure the required options 21 AEQ PHOENI X ALI O NOTE Calls cannot be made when the illumination of the key is flashing This is indicating that there are either issues with the physical link or the registration process with a SIP server when in PROXY SIP mode Communications can be established once the key stops flashing Please read chapters 4 2 2 4 2 3 5 3 2 and 5 3 3 for more details f o 3 7 2 AUTO key AUTO This key opens a menu where several automation options are available that simplifies the operation of the unit and automatic call management The adjustments made and displayed apply to the currently active channel only The AUTO button will be illuminated whenever at least one of the options is activated 3 7 2 1 SmartRTP AutoAnswer First we can select whether to activate or not the SmartRTP function available only in RTP mode that allows as explained before to signal call establishment and hang up from a single end of the call without using higher level negotiation protocols such as SIP When the operating mode for the currently selected channel has been set to SIP this option is substituted by Autoanswer which
53. age may vary depending on the web browser The saved image has a resolution of 132x68 pixels 4 bit pixel C O 192 168 1 88 grabscreen Importado de Internet Screen capture detail 7 6 Status menu By means of IP Status menu you can monitor some statistical parameters regarding the connection status of IP channels Some of these parameters are transmission and reception buffers status Jitter lost packets Channel 0 corresponds to PROGRAM and Channel 1 corresponds to COORDINATION C D 192 168 1 88 index htm PHOENIX Y ALIO Portable IP Audio Codec MENU IP STATUS UPGRADE STATUS CHANNEL 0 IP STATISTICAL SETTINGS BufferSize FIXED SIZE 100 ms Jitter 0 ms MAINTENANCE Received 151 Late 0 Repeated 0 Out of order 0 Error 0 Not received 0 a 153 Not played 0 Not found 0 Empty buffer 0 size adjustments 0 inserted 0 discarded 0 Clock drift 0 CHANNEL 1 IP STATISTICAL BufferSize FIXED SIZE 120 ms Jitter 792 ms Received 5209 Late 0 Repeated 0 Out of order 0 Error 0 Not received 0 Played 5210 Not played 0 Not found 0 Empty buffer 0 size adjustments D inserted 0 discarded 0 Clock drift 0 IP Status screen detail 7 7 SNMP This unit can be remotely managed using SNMP Simple Network Management Protocol using one of the many client pieces of software available in the market even for free SNMP allows monitorin
54. ain number of destination IPs depending on the type of coding algorithm That parallel streams or replicas at nothing more than IP address port pairs where audio copies are to be sent normally When a contact is created or edited it is also possible to specify whether particular replicas use FEC forward error correction if it is enabled for that channel or disable it for certain streams because they use stronger links for example If the above list is empty the audio stream will be sent to the IP port specified when making the call main address in the contact 38 AEQ PHOENI X ALI O SA In order to send replicas a new advanced contact must be created first in the General agenda where a main IP address and port is specified and a list of additional replicas is provided In order to do that click on the Contacts button at the top menu bar select New Contact make sure that the agenda selected at the left column is General and proceed with the creation of the new contact ControlPHOENIx Mew Contact Select where ta stare the Mew contact General ak peet new contact rale aeq 54 ETE PHAMASTER Studio SSM Interface Advanced Phone 1 Phone 2 IP Port 172 26 93 55 5004 Give the new contact a name l e RIP _REPLICA 1 select RTP as the interface mode specify the main destination IP address and port 172 26 33 55 5004 respectively in this example and click on the Adva
55. and size of the receiving buffer and FEC parameters as a function of the IP network quality so we have the shortest delay while audio cuts are minimized or eliminated in poor quality networks see paragraph 4 4 of this manual in order to select the optimal buffer configuration depending on your application Return to the general configuration screen check that the selected coding algorithm in the green ENCODER area corresponding to that channel PROG or COORD is correct or otherwise click on Select codec to change it ENCODER Coding AEQ LO 354kbs Select 48KHz STEREO coder Decide whether you will use the advanced automatic connection options or not o SmartRTP Activate this option o Auto hang up Automatic hang up whenever audio packets are missed for a given time o Permanent call The device will do what s necessary to keep the call up even after a line drop or mains cut When SmartRTP mode is active it is recommended that this option is enabled in the calling end only ControlPHOENIX J PRG Advanced configuration ALIO kl Smart RTP connect mode Auto hang up if sync loss after BI seconds Permanent call Return to the individual codec control window and click on CAL button of the desired channel showing then the call screen a Enter the IP address of the remote unit either manually or getting it from the buttons ControlPHOENIX ta Calls L Equipment contacts EE
56. and the user will be notified through the OLED display that the remote operator has acknowledged the call for assistance This key is not following the active channel and provides access to the configuration menu of the ALIO This menu also contains more advanced or less frequently used used options such as Phoenix ALIO IP configuration communications parameters RTP SIP and NAT parameters test options loops tones etc or maintenance operations firmware version time date adjustment system reboot buzzer HEHU ETHERHET COMMUNICATIONS MAINTENANCE BACE 3 7 5 1 ETHERNET submenu The first available option is EIHERNE TT configuration By entering this submenu pressing on the encoder button when the option is highlighted we can access the IP configuration of the unit where we can specify whether it gets the configuration automatically DHCP ON requires a DHCP server active in the local network ETHERHET DHCP of F IF 192 168 1 BACK 24 AEQ PHOENI X ALI O lf DHCP is not used we need to select the second line in this list and where the current IP address is shown Click on the rotary encoder and this will open all the available fields for the correct IP configuration of the ALIO and allows the user to change the settings for the IP address mask gateway and DNS server IP COMF IG MSE Isis UNS BACKE There are two ways to modify each IP clicking on the rotary encoder when the IP w
57. are not just operate the controls to establish the desired configuration e The unit is now ready to be remotely controlled The unit won t provide any visual or acoustic indication for around 15 seconds after it is connected until the welcome screens appear This corresponds to the boot time of the internal processor and doesn t indicate any problem or failure in the audiocodec 6 3 Setting up a computer to control the unit Connect to the same network a computer with AEQ ControlIPHOENIX software installed with version 2 2 0 4 or higher Follow what s indicated in chapter 3 of the application s user manual Check that your PHOENIX ALIO is automatically discovered after launching the application according to chapter 4 1 of the software user s manual Accept the unit and if you dont find it please check that the network parameters of both the unit and the computer belong to the same network having in mind the default IP address of the unit IP 192 168 1 88 Mask 255 255 255 0 GWAY DNS 192 168 1 100 The codec is ready to be controlled when the individual codec control window appears showing OK label in PRG and CRD status areas and below the equipment s given name ALIO1 in the example below PHOENIX ALIO OK Gs not connected CH n OK not connected ALIO WIRI 1 pain 14 WRI 2 phxalio 15 56 AEQ PHOENI X ALI O IMPORTANT NOTE If more new codecs will be controlled in the same netw
58. associated rotary encoder with pushbutton and an activation routing key ON If the rotary encoder is turn the mix gain to the selected bus is altered for that input between mute infinity and 18 dB The level bar at the display shows an indication of this level The ON buttons allows the user to select which bus the associated input will be sent to When it is off pressing it will turn the button RED illuminated indicating that the input is sent to PROGRAM transmission channel One more pressing and the key will be illuminated in GREEN indicating that the channel is sent to COORDINATION transmission If it is pressed once again it will be off again and the input will only be available locally in the CUE pre listen bus The selection is cyclic for each input PROG CUE COOR When several inputs have their ON button illuminated in the same color they will be mixed each source according to the selected mix level to the selected bus PROG COORD or CUE When the encoder button is pressed access to the associated input s configuration menu is provided Here the user can configure the corresponding gain equalization bass treble activation of the 12V Phantom supply and in the case of MIC4 LINE input the user will also be able to select which input is used microphone 4 or the line input at the right panel WARNING DO NOT turn Phantom power on or off for a microphone unless the Microphone Gain is set to minimum Dependi
59. aving and loading configurations In the MAINTENANCE section in the lower part of the screen you will see the CONFIGURATION MEMORY option from which you can save the current configuration of the unit by means of DOWNLOAD button or load a configuration previously created and saved by selecting the corresponding file and pressing then the Save configuration button The extension of the files used in this process is AFU The Reset configuration button allows you to restore the default configuration of the unit gt C D 192 168 1 88 index htm Ode PHOENIX Y ALIO Portable IP Audio Codec MENU SNMP UPGRADE ST ATUS Download WIB SETTINGS SYSTEM REBOOT MAINTENANCE CONFIGURATION MEMORY DOWNLOAD configuration Select file Seleccionar archivo Ning n archivo seleccionado Send config Configurations screen detail 69 AEQ PHOENI X ALI O 7 5 Screen capture The unit allows you to capture the screen displayed in the multifunction screen at any moment in order to save it into a PC in BMP image format This function may be useful in order to prepare documents or report problems or doubts to SAT In order to capture a screen open the web browser and write the following text in address bar http lt unit_IP_address gt GrabScreen A new screen will appear showing the image Place the pointer on the image press the right mouse button and select Save Image as the mess
60. ay will be different than when its done through the Internet See more details in section 4 3 of this manual The rest of options to be configured are o FEC mode this option allows you to configure whether FEC Forward Error Correction is used or not there is a trade off for a bigger binary rate See section 4 4 36 AEQ PHOENIX ALIO o Local media port this option allows you to configure the value of the IP port selected to transmit audio at origin over IP Minimum value 1 024 Maximum value 65 534 Default values 5004 PROG and 5008 COORD o Adaptive Fixed and Adaptive buffer max Fixed buffer length this option allows you to configure the type and maximum size of reception buffer See section 4 4 o Symmetric RTP this option allows you to force the local unit to send audio to the same IP and port from which it is receiving audio The destination port specified when making the call will be ignored when we receive packets from the remote equipment This option will allow you to connect to equipment with unknown IP and or port because it s behind a router with NAT for instance Each unit will send audio to the Local media port of the remote equipment automatically thanks to the SIP signaling protocol That signaling also accomplishes coding profile negotiation and call establishment release from any of both sides of the communication once the remote equipment has been identified by its IP address and reached 4 2
61. be always available and required even if the channel is configured in SIP mode ATF BUFFER fixed size SIZE Z ms SYMMETRIC PTF LOCHDL FORT DD FEC off The parameters that can be configured in this menu are Receiving BUFFER type FIXED or ADAPTIVE size as well as its SIZE in milliseconds A fixed size buffer is recommended whenever possible unless the opposite is indicated by AEQ for a certain application The size to select depends on the network performance in particular on the variation of the delay in packet delivery 25 AEQ PHOENI X ALI O jitter The size specified for the buffer must be higher than the maximum variation Note that a certain network may have a long delay but quite constant low jitter as is the case in good satellite links In this case the buffer size can still be short Large buffer sizes results in unnecessarily long added delays We recommend to start with a value around 100 ms that the user should increase only in case that drops or occasional losses of synchronization is experienced Such occasional loss is most likely due to that the network jitter at times may be larger than those 100ms In order to diagnose network jitter we recommend that the IP statistics available in the Web Server are used read section 7 6 of this manual Activation or deactivation of SYMMETRIC RTP mode that sends the transmitted audio stream to the same IP and port that the unit is receiving from ind
62. bing a contact can be modified in the Call Book that can be accessed from a codec individual control window through the Contacts option in Configuration see section 5 1 7 of AEQ ControlIPHOENIX user s manual In order to call a same contact using different IP modes as defined in INTERFACE drop down menu different contact entries must be created General E Ethernet p C emo You can access the IP configuration submenu for PROXY SIP mode by clicking on I F Setup button and that it is explained in section 6 1 4 2 of AEQ ControlPHOENIX user s manual 35 AEQ PHOENI X ALI O In SIP Parameters submenu you can find ControlPHOENIX Configuration PRG SIP parameters ALIO Local uri Proxy SIP Proxy SIP Accounts User name phxalio_xxx displayname Phoenix Channel 1 v J Manage Proxy SIP Accounts Proxy Provider Authentication AEQ v Password oe 4 Manage Providers Realm sip aeq es User Name enables you to edit the name of the unit and how it will be reflected in the diverse internal menus of the unit For a start we recommend you not to change the configured default User Name phxalio_231 for instance Display Name editable name This is the public name of the equipment that will be used in SIP server so you can recognize the equipment with this identifier externally
63. call in PROXY SIP mode nnnnannnnnnnnnnsnnnnnnnnnsnrnnsnnnnnsnrnesenenne 59 6 5 2 1 Ending an IP call in PROXY SIP mode 62 6 5 2 2 Receiving and accepting an IP call in PROXY SIP mode 62 6 5 3 Establishing an IP call in DIRECT Glbmode cc ccccccceeeeeeeeeeeaeeeeeeneeeeenaees 63 6 5 3 1 Ending an IP call in DIRECT SIP mode cccccecceceeeeeeeeeeeeeeeaeeeeeens 65 6 5 3 2 Receiving and accepting IP calls in DIRECT SIP mode 00 65 7 CONTROL TERMINAL OVER WEB BROWSER ccccssesseeeeseenseeeseeeesseeeseeesnseeseeenseeseees 67 7 1 Upgrading system firmware cccccseeeceeeeeeeeeceeeeeeeseeeeeeseeseeseeeeeeeseeeeeesaaueeessaneeesaeeeeesaaes 67 7 2 Configuring the MAC address associated with the Ethernet interface ccceees 68 7 3 Technical Assistance Service and on line manuals ccccecceeeeeeeeeeeeeeeeseeeeeeaeeeesaees 69 7 4 Saving and loading configurations ccceccccceeeeeeeeeeeeeeeeeeeesaeeeeeeeeeeeeeseueeessaeeessaeeeeesaaes 69 eene et 70 LO UAVS WN EE 70 Lr O spo reste cneeesucosey oles E E E cove nue p E so aseeaedeenavedats eadsneay 70 7 8 Remotely rebooting the equipment ce cccceeccceeeeeseeeeseeeeeseeeeeseeessaueeesaeeeseeesauseeseees 71 SG TECHNICAL SPECIFICATIONS eege 72 SAET VAY E 73 APPENDIX A GENERAL CHARACTERISTICS OF ENCODING MODES seen 74 APPENDIX B PROTOCOLS ASSOCIATED WITH IP COMMUNICATIONS eessen 76 B1 Circuit s
64. ccesses or point to point CONNEC UOS ee E E E A E S E 87 PANIC ALON NO AN ME 87 el elteren NOC EE 87 APPI AION MO AN er e E EE E Ee 87 Application note E EE 87 Application Note ANS EE 87 Application note AN5B EE 87 APPENDIX E ADDITIONAL INFORMATION cccccesseeeccesseeeeenseeeeenseecenseeseoeseessonneeeses 88 4 AEQ PHOENI X ALI O 1 INTRODUCTION 1 1 General description AEQ PHOENIX ALIO is a stereo IP audiocodec for mobile applications It is easy to configure and operate and integrates a digital mixer with 4 analog inputs It features independent bass and treble control for each input adapting the characteristics of each speaker s voice or correcting the defects of external signals It has been specifically designed for sports reporting applications but it has also been optimized to make it easy to use in the most varied scenarios even musical events Its IP connectivity allows the user to choose among a wide variety of connection modes dedicated networks DSL cable modem fiber optic WiFi Wig MAX Inmarsat satellite etc The application notes listed at the end of this manual provide information about how to operate the unit in each case AN 5B stands out among them describing a simple and effective way to connect the audiocodec to 3G and 4G networks as well as to WiFi hotspots through a router and modem that can be provided by AEQ or obtained locally PHOENIX ALIO is optimized for OPUS encoding a
65. ction 4 3 Ending an IP call in PROXY SIP mode In order to hang call just click on CALL button corresponding to that channel in one of both codecs individual control window A confirmation message will appear and the call will be disconnected after acceptance Receiving and accepting an IP call in PROXY SIP mode If the IP interface is correctly configured and automatic answer mode is OFF when you receive a Call The unit and the application will provide acoustic warning This can be disabled for the unit at Configuration gt Miscellaneous gt Buzzer and test The CALL button red LED corresponding to that channel at the individual codec control window in the remote control software corresponding to the unit that is receiving a Call will blink at the same time to warn the user In addition if Autoanswer option is not active an incoming call window will appear showing the URI identifier of the caller unit ControlPHOENIX Incoming Calls AEQ Dat From 04 10 2012 08 53 25 sip phoenixMaster sip aeq es internet 172 26 5 59 The call will be accepted by clicking on the individual codec control window CALL button corresponding to that channel or alternatively on the Accept button in the incoming calls window The screen will show the status of the call o CONNECTING o SYNCHRONIZING o CONNECTED NO_SYNC NO_ DATA Verify that the SYNC LED beneath the
66. ction of e Aencoding algorithm among those supported by the unit for RTP calls e A encoding profile list of prioritized preferred modes used when negotiating calls with other units for SIP calls 22 AEQ PHOENI X ALI O The same way as the other function keys except HELP and MENU the selection is accomplished for the currently active channel PROG or COORD the latter only if license is activated PROG CHAHHEL PROG CHAHHEL RTP MODE SIP MODE sl Cd T Eg m nin nia m T AN H M E MUSIC MUSIC ST BE B As can be noticed in the above sequence of screen captures when in RTP mode the encoding algorithms are presented grouped by families OPUS G 722 AEQ LD PCM G 711 etc and several particular modes are available within each family A u law mono stereo different bitrates sampling frequencies etc When a particular mode is selected successive calls will be made using that algorithm and the remote codec must be configured just in the same way unless SmartRTP mode is activated in this case the encoding algorithm will be automatically notified to the remote audiocodec In SIP mode however a list of encoding profiles is presented Each profile is basically a list with a name including one or more precise encoding algorithms and in order of preference When a unit calls another in SIP mode it proposes this list and the first one supported by both audiocodecs will be the one finally adopted
67. digits per number and confirming each number with the NAVI Ch button AEQ PHOENI X ALI O 50 For example If you want to call to IP 192 26 5 12 just type 1 9 2 NAVI Ch button 0 2 6 NAVI Ch button 0 0 5 NAVI Ch button 0 1 2 NAVI Ch button to finish e Go to next line PORT and select the destination port either with the encoder they change in steps of 4 or typing it always with 5 digits e Select the CALL option or press the OK key again e Repeat the audio and mode configuration process in the other end As SmartRTP option is active the other codec will automatically connect as soon as it starts receiving RTP traffic and so there will be no need to dial IP addresses or ports accept the incoming call or even configuring the encoding algorithm e The OLED screen displays the call status as well as the destination IP and port o CONNECTING The ORT key will blink when in this status CUE CUE CUE COMHECTING Jet Vee Fia CAN CODEC AE Q LO MIL HIC MICE HICA HFA HEE OWT o CONNECTED NO SYNC NO DATA When the call has been successfully established data is received but synchronization to it is not possible or no data at all is received respectively The OK key will remain steadily illuminated CHE CHE CHE CUE CUE CUE CUE CUE CUE LOHHE TEL COHHECTED HO SYHC LOHHEC TED HO DATA 192 168 1 55 Shee 192 1
68. e PROG and COORD labels located between the vumeters When a call or communication is established in the channel that is not active the label will blink while the other remains steadily lit THI HDH gt TOT TTT TE TLL Li COORD PROG COORD TT IHR E III HUTT 3 3 Vumeters c The unit is equipped with two stereo vumeters From above the first one is indicating the level of the TX signal The second indicates the level of reception RX If the unit has the second optional coordination channel activated the vu meters will follow the NAVI Ch encoder and will display the levels corresponding to the active channel and with the labels PROG and COORD between the vumeters and as explained above Each of the 4 vumeters consists in 2 bars of 20 LEDs each From right to left the three first LEDs are red followed by four orange coloured ones The remainders of the LEDs are green The scale applied has higher resolution around the transition between green and yellow colors Please note that the presence of audio is denoted at 36 dBV when the first green LED is illuminated Transition from green to yellow is at 0 dBV from yellow to red is at 7 dBV and when all 3 red LEDs are on displayed level corresponds to 13 dBV or higher 46 0 Z9 20 i6 14 12 10 E amp 4 d 1 D 1 kd r 4100 13 AEQ PHOENI X ALI O 3 4 Alphanumeric keyboard and call buttons o Standard numeric keyboard with
69. e ORT button The numeric keyboard will become illuminated in a color that depends on the active channel red PROGRAM or green COORDINATION Using the NAVI Ch encoder select whether you want to call from an agenda contact CONTACT LIST or manually dial an URI If manual mode is selected go to the URI line and fill the destination name as it you were typing a SMS on a mobile phone The URI must be specified according to one of the following formats lt dest_URI gt L phoenixMaster lt dest_URI gt lt PROXY_domain gt L phoenixMaster sip aeq es lt dest_URI gt lt PROXY_IP gt le phoenixMaster 232 168 1 2 lt dest_URI gt lt PROXY_domain gt lt SIP port gt L phoenixMaster sip aegq es 5060 O O O 0 Select the CALL option or press the OK key again Accept if necessary the call in the other end see 5 3 2 2 The OLED screen displays the call status as well as the destination URI address o CONNECTING The OK key will blink during this time o CONNECTED NO SYNC NO DATA When the call has been successfully established data is received but synchronization to it is not possible or no data at all is received respectively The OK key will remain steadily illuminated If status is NO SYNC or NO DATA and auto hang up option is enabled the call will be rejected after the defined time and the OK key illumination will turn off Once connected with the remote end verify that the vume
70. e established experiment with the different available modes in case of problems and check if the results are better with some of them e LOWEST generates a 40 higher binary rate and produces a 5 5ms additional delay e LOW generates a 50 higher binary rate and produces a 3 75ms additional delay e MIDDLE generates a 66 higher binary rate and produces 225ms additional delay e HIGH duals the binary rate producing 125 ms additional delay Adaptive Fixed you can set up the reception buffer as adaptive or fixed In the first case its size will vary according to the network transmission quality In fixed mode its size will be steady according to manual configuration Adaptive Buffer Max Fixed buffer length this is the maximum size of the reception buffer When it is defined as adaptive Phoenix ALIO will start to shorten it from this value as the network s transmission quality allows If it is defined as FIXED this max value will remain as the buffer s size won t be varied during the connection This value must be set in milliseconds The longer the buffer is packet misses will be less likely but base delay will also be longer especially if the buffer is set to FIXED mode In order to help you select the best option for each application we recommend to use a Fixed buffer with a low value around 100ms in applications where optimal audio quality is the main concern mainly when using PCM modes in suitably sized networks I
71. e name of the unit and how it will be reflected in the diverse internal menus of the unit For a start we recommend you not to change the configured default User Name phxalio_231 for instance Display Name editable name This is the public name of the equipment so you can recognize the equipment with this identifier externally to the system e You can find the NAT mode selection at NAT Traversal submenu NAT Traversal is a set of tools used by the equipment in order to surpass the NAT Network Address Translation performed by some routers We can select among several modes depending on the kind of network the unit is connected to Phoenix ALIO offers a total of six different operating modes when traversing devices with NAT routers firewalls Each one of those modes is suitable for a different scenario For instance when the units that are establishing communication are in the same local network the internal working way will be different than when its done through the Internet 37 AEQ PHOENI X ALI O See more details in section 4 3 of this manual e The rest of options to be configured are o FEC mode this option allows you to configure whether FEC Forward Error Correction is used or not there is a trade off for a bigger binary rate See section 4 4 o Local media port this option allows you to configure the value of the IP port selected to transmit audio at origin over IP Minimum value 1 024 Maximum value
72. e selected and we can move the knob to select the next line or go BACK In order to adjust time select the second line and proceed just in the same way The clock is of the 24h type If some changes have been made when selecting the BACK option or pressing the ESC DEL key they will be automatically applied e TESTS We can find some test functions under this menu such as activation of audio loops for each channel and test tone insertion that substitute the corresponding input for inputs MIC1 and MIC2 that we can route to the desired channel or CUE in order to make level adjustments check audio connectivity etc The first and second lines allow for the activation of an audio loop in PROGRAM and COORD channels This loop is linear no compression Audio entering the TX bus will be reflected back in the receive RX bus This will also be shown in the corresponding vumeters and can be useful to measure audio performance adjust levels check connections etc NOTE LINEAR LOOP cannot be activated when the corresponding channel is in call mode currently 2 AEQ PHOENI X ALI O The third line option enables disables the substitution of MIC1 input by a 1 kHz tone at 20 dBV level adjustable between inf and 18 dBV by means of the MIC encoder The mentioned tone will be sent to the bus where MIC1 is routed and the microphone will remain muted if it is connected The fourth line enables disable
73. e want to change is highlighted will allow us to change the value for each of the four sections of the IP address between 0 and 255 We can increment or decrease by turning the rotary encoder Clicking on the rotary encoder when we have the desired value selected will set the value and jump to the next section and so on until we have completed the IP address that is required It is also possible to enter the numbers with the numeric keypad of the ALIO always completing the 3 digits For example if we want to type 6 we should type 0 0 6 then confirm each number clicking on the rotary encoder The default IP address for PHOENIX ALIO is 192 168 1 88 If any of the IP or mask is altered either by selecting DHCP or manually changing it a warning saying that the unit will reboot to apply the changes will appear when leave the ETHERNET submenu 3 7 5 2 COMMUNICATIONS submenu Advanced parameters related to communication protocols can be configured here either basic ones such as RTP configuration or when the channel is in SIP mode parameters related to this protocol or to NAT Traversal tools COMMUHICATIOHS SIF HAT BACK e RTP Generic parameters that are always necessary for IP audio transmission using standard UDP type RTP Real Time Protocol are configured in this menu Note that this protocol is always used even when a higher level protocol such as SIP is active so its configuration will
74. eeded it is provided by transport layer protocols such as TCP Transport Control Protocol Reliability over TCP is obtained through the use of retransmissions Real time applications such as an audio link with the timing requirements inherent in the information contained in the link do not offer any useful guarantee Since the data that are not received and whose retransmission is requested of the sender by the receiver will in most cases arrive out of order they will end up as useless information that will have served only to overload the network For all these reasons the protocol selected to serve aS a communication substrate in real time applications is UDP UDPDatagram 78 AEQ PHOENI X ALI O Transport over IP protocols independently of the reliability they offer add new functionalities to the basic ones provided by IP such as packet numbering to facilitate on the receiving end the detection of losses although not their correction and of disorder in the information received and the advent of the port concept as an identifier of different logic connections over the same IP interface For complete information on IP protocol we recommend consulting http tools ietf org html rfc791 http www iana org assignments port numbers B2 1 IP addressing An IP address is a number that logically and hierarchically identifies an interface of a device in a network that uses the IP protocol The format used is X X X X where each X
75. ent SIP 45 AEQ PHOENI X ALI O 4 6 Ethernet Port configuration The Ethernet config menu allows you to configure the IP parameters of the Ethernet interface in the unit ControlPHOENIX Ethernet contig A i LIO Ethernet module 1 Enable DHCP IP Address ER Subnet mask 255 Gateway IP 192 DNS Server The parameters to be configured are Enable DHCP enables the activation or deactivation of the automatic configuration of IP addresses masks and gateways There must be a DHCP server in the network the unit is connected to in order to make this option work When Enable DHCP is validated the following parameters will be filled automatically when Enable DHCP is not validated you will be able to change them manually IP Address valid IP address associated with that interface Subnet mask valid subnet mask associated with that interface Gateway IP valid gateway or network gateway address associated with that interface DNS Server IP address of the external addresses resolution server valid in the geographic zone where codec is placed or internal server inside the local network authorized to translate alohanumeric URL identifiers into IP addresses Once those parameters are configured and after pressing the Apply button a confirmation window will appear After confirming the equipment reboots and the communication re establishes in approxi
76. ep the call up even after a line drop or mains cut o Apply audio profile to incoming calls allows you to filter the SIP calls depending on the encoding profile of the receiver 60 AEQ PHOENI X ALI O ControlPHOENIX J PRG Advanced configuration ALIO Ti Autoanswer Autoanswer Number URI Leave blank For any L Auto hang up if sync loss ve after bo seconds Permanent call Apply audio profile to incoming calls Return to the individual codec control window and click on CALL button corresponding to that channel showing then the call screen ControlPHOENIX L ALIO Call on PRG gt Call to gt phoenixMaster Channel interface SIP Proxy based Calls LL Equipment Contacts BR P 192 168 1 68 5008 Channel 1 15 01 2010 09 38 27 00 02 35 192 168 1 68 5008 Channel 1 gt 15 01 2010 09 33 59 00 00 07 Enter the IP address of the remote unit either t calls EE mengen edit manually or getting it from the buttons By clicking here the last URI Calls the URIs in the Equipment Contacts book or the available IP addresses are shown respectively but only those with formats compatible with channel and communication type It is mandatory that the called unit URI is specified in any of the following formats adequate for Proxy SIP communications o lt equipment s_name gt for example phxalio_231 or phoenixMaster o lt
77. ependently from the setting of destination port this option doesn t appear when SmartRTP mode is active as it is no longer necessary We must also specify the local audio port here port where the unit expects to receive audio RTP traffic at This LOCAL PORT is 5004 by default for program channel and 5008 for the optional coordination channel It must be a multiple of 4 and can be typed in with the numeric key pad always with 5 digits or modified with the NAVI Ch rotary encoder Last we can define here whether to use FEC Forward Error Correction or not If enabled the protection level and consequently increased overhead must be specified among the 4 available options LOWEST LOW MID HIGH Please refer to section 4 4 of this manual for more details about FEC SIP Next option in the communications menu corresponds to SIP configuration This option will only appear when the currently active channel has been configured in SIP mode with the corresponding key see 3 7 1 SIP PROM off PROVIDERS aes 15 ACCOUNT lu BACK Hl The parameters that can be configured in this menu are First we need to select whether a PROXY server is used or not to differentiate between PROXY SIP and DIRECT SIP modes In case we activate it a new option to be specified will appear in the menu The PROVIDER should be choosen among the preloaded listed options It is possible to insert a new provider by chosi
78. er in the connection RSR MLCT onus ab op our Geroch daer RTP Header For complete information on RTP RTCP protocol we recommend consulting http tools ietf org html rfc1889 http tools ietf org html rfc1890 http tools ietf org html rfc3550 http tools ietf org html rfc3551 http tools ietf org html rfc371 1 80 AEQ PHOENI X ALI O B3 1 Default PHOENIX ALIO configuration PHOENIX ALIO is an IP audiocodec that operates by using RTP over UDP in IP version 4 By default PHOENIX ALIO is supplied from the factory with the following IP ports defined 5004 for RTP and the next one 5005 in this case for RTCP for the PROGRAM channel and 5008 5009 for the COORDINATION channel if licensed The RTP port values can be modified from its internal menu and RTCP ports will be automatically assigned accordingly B4 SIP protocol Session Initiation Protocol SIP is a protocol developed by the IETF MMUSIC Working Group with the intention of establishing the standard for initiating modifying and ending interactive user sessions involving multimedia elements such as video voice and instant messaging SIP is used simply to initiate and terminate voice and video calls Once the communication is established the exchange of voice video information is conducted only over RTP One of the objectives of SIP was to contribute a set of processing functions to apply to calls and capacities present in the public switched telephone n
79. etwork Thus it implemented typical functions that a common telephone terminal offers such as calling a number making a telephone ring when called hearing a dial tone or busy tone The implementation and terminology in SIP are different SIP requires proxy servers and register elements to give a practical service Although two SIP terminals can communicate with each other without the mediation of SIP infrastructures through the use of URIs of the name I P address type which is why SIP is defined as a point to point protocol this approach is impracticable for a public service because of the problems inherent in IP addressing where obtaining static public addresses is nearly impossible and extremely costly To simplify the operation of the unit AEQ offers at no additional cost the services of its 2 own SIP servers one of them working as main server and the other one as backup server although it cannot guarantee its operation 100 of the time nor be held responsible for the inconveniences that this may produce for the end user The unit leaves the factory preconfigured with the parameters required to work with the resources of any of these 2 SIP servers SIP makes use of elements called proxy servers to help route the requests toward the user s current location authenticate users to give them service enable call routing policies to be implemented and contribute added capabilities to the user SIP also contributes register functions
80. eueesseessueesseeenaass 33 d2 een Biel 34 4 2 1 RTP Foint to Point RAW EE 34 a PROA EE 35 ER EE 37 4 2 4 Sending audio to multiple destinations Broadcast Multicast and Multi unicast 38 OANA TRAVERSAL E 41 4 3 1 Operation without NAT OFF there is no NAT cccccceccessseeeeeseeeeeeeeeeeeeaeeeees 42 4 3 2 Manual NAT MANUAL router Configuration cccccccccessseeeeeseeeeeeeeeeeeeaeeeees 42 4 3 3 AUTO 1 local network audio 43 4 3 4 AUTO 2 local NetWork audio 43 4 3 5 AUTO 3 audio over internet 2 eecccccceeeeeeeeeeeeeeeeeeeeeseeeeeseeeesseeeeeesaaeeeessaaeees 43 2 AEQ PHOENI X ALI O 4 3 6 AUTO 4 audio Over internet 0 0 ceecceccseeeeeceeeeeeeeeeeeeeseeeeeeseeeeesseeeeesaeeeeesaaeeees 44 4 4 FEC modes and reception buffer configuration ceccccceeeceecaeeeeeeeeeeeeeaeeeeeeaeeeeesaaeeees 44 4 5 Coding algorithm selechon ccccceeceecseeeeeeeeeeeeeeeeeeeeaeeeeeseeeeeeesaeeeesseeeesaeeeeesaeeeesaaeeees 45 4 6 Ethernet Port configuration 1 0 ce ecceceeeeeeeeeeeeeecaeeeeesaeeeeeeaaeeeesaaeeeesseaeeeeseaeeeesaaeeeesaeeees 46 4 7 SNMP Configuration ccccceecccccseeeeecaeeeeeeeeeeeeeeeeeeeseeseeeseeeeeeseeeeesseseessaaeeesseeeeesaaeeees 46 5 QUICK USER S GUIDE LOCAL OPERATION 1 00 cet eeeccseeeeeeeeeeneeseeeeeeeeeeeeeseeeeeeeeeeeenneeeeees 49 5 ll COMMECHING TNE ln LEE 49 Ds Te do Power SUD DIY EE 49 e E ele EE 49 5 1 3 COMMUNICATION
81. evice Version Upgrade MAINTENANCE CPU 05 00 11 03 15 FPGA 05 54 21 09 15 DSP 03 33 09 30 15 Select file Seleccionar archiva Ning n archivo seleccionado Upgrade Firmware upgrading screen detail 8 Check to see whether the versions displayed are the same as the firmware that is currently in effect If they do not match upgrade the firmware as indicated below 9 Select the module you want to upgrade in Upgrade column NOTE Each upgrading file is specifically designed to upgrade a specific module within the unit CPU DSP or FPGA 10 In Select file enter the access route to the upgrade file containing the new firmware version using the Seleccionar archivo button 11 Press the Upgrade button in the lower part of the screen 12 Wait for on screen confirmation that the operation has been successfully completed 13 In the Internet browser go to the UPGRADE section and ensure that all the firmware versions installed in your codec are now the correct ones 14 Power the unit down 7 2 Configuring the MAC address associated with the Ethernet interface From this menu the MAC address associated with the Ethernet interface can be edited because of the consequences this action could have the addresses should only be edited if the codec use situation requires it The editing should be performed by highly qualified personnel or under the supervision of AEQ authorized technical services and always in pos
82. f making an RTP call is greatly simplified When operating Phoenix family of AudioCodecs with this mode activated it is only required that the caller launches the call The remote equipment will automatically answer and send its audio stream to the callers IP and port Further the AudioCodec will also detect and automatically select the encoding algorithm that the calling unit is using to initiate the communication The call does not need to be manually accepted and the hang up event from any end will also be signaled to either unit NOTE The illumination of the key will flash whenever the associated channel is unavailable because the Ethernet link is down Once the flashing of the key stops communications can be initiated Please refer to chapters 3 7 2 4 2 1 and 5 3 1 for more details SIP mode N ACIP compatible SIP Session Initiation Protocol is an alternative to RTP for making a call The SIP key configures the unit so all communications made with the currently active channels use this signaling protocol compatible with other codecs following N ACIP standard from EBU It can be used directly between audiocodecs DIRECT SIP or taking advantage of an external Proxy PROXY SIP Its function is to maintain a database of all registered codecs with their IP addresses and listening ports making it possible to establish a connection between audiocodecs located in different networks This allows the user to forget about the IP
83. f the received audio quality is as expected and the network allows for it you can continue adjusting the buffer to lower values in order to minimize delay until you find that audio is compromised as the buffer size reaches the network maximum jitter value At this point just increase the buffer a little bit to have some margin In high quality PCM connections you can start using highest quality modes 48KHz 24 bits mono or stereo only if required and if you can t obtain the desired quality and or stability no noises present and good delay you can lower quality progressively until for example 16 bit CD quality audio On the other hand for applications where lowest possible delay is the main goal but transparent audio is not necessary for example in voice connections with commentators it is better to select the Adaptive Buffer mode starting from a 1000ms maximum size approx If the network is not too bad the unit won t increase the buffer to highest values from the network s jitter value and it will try to minimize delay continuously Please not that if the network has very variable delay the adjustments required to increase or decrease the buffer size can produce noticeable artifacts in the received audio so this method is not recommended for PCM modes where maximum quality is required in this case a fixed buffer setting is preferred as stated above 44 AEQ PHOENI X ALI O 4 5 Coding algorithm selection See sectio
84. g of the status of several pieces of equipment from very diverse manufacturers and natures as well as elaborating reports generate e mail alerts etc 70 AEQ PHOENIX ALIO ears EH In order to add and equipment to the list of units managed by the client it is necessary to have access to its MIB file Management Information Base that describes its SNMP capabilities alarms it can generate accepted commands manufacturer information etc The MIB file corresponding to the unit can be downloaded from the Web interface without installation of any additional software In order to do so in the MAINTENANCE section you can access the link Download MIB under the SNMP section D 192 168 1 88 index htm PHOENIX Y ALIO Portable IP Audio Codec MENU SNIP UPGRADE ST ATUS Download WIB SETTINGS SYSTEM REBOOT MAINTENANCE Reboot CONFIGURATION MEMORY DOWNLOAD configuration Select file Seleccionar archivo Ning n archivo seleccionado Send configuration Reset configuration SNMP screen detail If you follow that link the text file will appear Now you just need to right click on it and select Save as and browse a suitable destination folder see the manual of the selected SNMP client For more information please consult section 4 7 of this manual and section 6 5 1 of AEQ ControlIPHOENIX application manual 7 8 Remotely rebooting the equipment Sometimes
85. guaranteed with Internet Explorer running on Microsoft Windows operating system By default user and password is aeq IMPORTANT NOTE the recommended order for upgrading is MICRO CPU DSP and FPGA The process is iterative To upgrade the firmware you must follow the steps detailed below 1 From MENU gt ETHERNET check the IP address associated with the Ethernet interface 2 Power down the PHOENIX ALIO 3 Connect PHOENIX ALIO to the PC from which you are going to perform the upgrading process using a crossed cable 4 Power up the PHOENIX ALIO 5 Open the Internet Explorer web browser and in the address bar enter HT TP lt IP address obtained in point 1 gt Press ENTER and the main screen will be displayed e D 192 168 1 88 index htm PHOENIX Y ALIO Portable IP Audio Codec MENU MAIN PAGE UPGRADE STATUS SETTINGS MAINTENANCE Main screen detail 6 To upgrade the codec click on the UPGRADE option 67 AEQ PHOENI X ALI O s e f j f A i fw 4 F F a Ti i 4 j ZG 8 i f B ss d 4 i 7 T fi Zi i 4 ff E f i d Ir i H d Ji 1 E i fie Z y AA AH a fi 7 q SERA i fj 7 7 A user ID and password are requested by default both are aeq After you have correctly entered these two items the firmware upgrading screen will be displayed C D 192 168 1 88 index htm PHOENIX X ALIO Portable IP Audio Codec MENU UPGRADE UPGRADE STATUS FIRMWARE INFORMATION SETTINGS D
86. h the SmartRTP option activated this option can be found under the AUTO menu as it will be adequate for most cases together with the OPUS family of encoding algorithms When operating with audiocodecs from other manufacturers you can use either IP mode without SmartRTP almost every codec out there can operate in RTP mode or alternatively a SIP based mode When the IP address of the counterpart unit is not known mobile connections dynamic IP etc the PROXY SIP mode is recommended as it relies on an external server to resolve IP addresses On the other hand if IP and ports are well defined it is better to use DIRECT SIP Select any of the N ACIP recommended encoding algorithms such as G 22 or MPEG 1 2 depending on the quality required acceptable delay and available bandwidth 5 3 1 Establishing an IP communication in RTP mode using SmartRTP Check that the unit is ON Check that the RJ45 cable is connected and latched Check that the amber LED integrated in the RJ45 port is flashing Verify the status of the communications interface Activate the IP button which will remain steadily illuminated If it blinks there is some problem with the Ethernet connection and the unit won t allow you to make calls e Press the AUTO button and make sure that SmartRTP option is activated e Establish the desired audio configuration inputs routing mix level MIC4 LINE IN mode and outputs e lf the coordination c
87. hand we need to cut the call from a receiver only that with transmission enabled will make the others hang up even if it is not the main receiver Please read the application notes published by AEQ regarding IP connectivity for more information on IP communications in particular scenarios 4 3 NAT TRAVERSAL NAT Traversal is a set of tools used by the equipment in order to overcome the issues caused by NAT Network Address Translation performed by some routers We can select among several modes depending on the kind of network the unit is connected to Phoenix ALIO offers a total of six different operating modes when traversing devices with NAT routers firewalls Each one of those modes is suitable for a different scenario For instance when the units that are establishing communication are in the same local network the internal working way will be different than when it s done through the Internet Four of the six modes are automatic AUTO 1 AUTO 4 another one is manual MANUAL router configuration and the last one OFF there is no NAT is used when no devices with NAT are crossed the unit is in a local network or connected to the Internet with a single workstation router In automatic modes the unit tries to find out its public IP and ports without the user help while in manual mode the unit gets those data directly from user and user gets it from network administrator Due to the technical complexity inherent in
88. hannel license is activated select the channel we want to call from by means of the NAVI Ch button The corresponding legend will get illuminated between the vumeters PROG or COORD e Configure the coding algorithm according to the desired quality and network capabilities by pressing the CODEC button Choosing one of the OPUS modes included in the unit is recommended depending on the communication needs voice mono music stereo music and network connection quality e Press the OK button The numeric keyboard will become illuminated in a color that depends on the active channel red PROGRAM or green COORDINATION e Using the NAVI Ch encoder select whether you want to call from an agenda contact CONTACT LIST or manually dial an IP Port If manual mode is selected go to the IP line and fill the numbers moving the encoder and confirming each one by means of its pushbutton PROG CALL COMTACT LIST IPF 172 276 355 205 PORT 56d CALL CAHCEL PROG CALL CONTACT LIST IF 159 2 165 1 02 PORT L ps CALL CAHCEL PROG slk LI Ee 633 823 d RT SAd CALL CAHCEL PROG CALL COMTACT LIST IF 197 168 661 656 PORT 2004 CALL CAHCEL PROG CALL CONTACT LIST IP 192 166 CAHCEL FROG CALL NIK LIST e DA DZ d et a ee CALL CAHCEL PROG CALL COMTACT LIST IF 137 163 1680 07 PORT 2004 CALL CAHCEL Alternatively you can directly dial the IP address always typing 3
89. he HP1 and or HP2 outputs at the right side of the unit e f required connect the line output to the XLR connectors at the right side of the unit labeled as OUT L amp R 5 1 3 Communications interface e Connect an Ethernet cable CAT5 or better finished in an RJ45 10 100 BT to the LAN connector provided at the unit s back panel The selected cable must be straight when the connection is made from the unit to a communications device switch router For more information about the pinout of this port please check section 2 2 2 of this manual 5 2 Turning the unit on e Once the unit is connected to the power supply through its adapter the OLED screen will turn on after around 15 seconds showing AEQ logo and the audiocodec model name e Check that audio routing and levels are correct if they are not just operate the controls to establish the desired configuration e The unit is ready to be used The unit won t provide any visual or acoustic indication for around 15 seconds after it is connected until the welcome screens appear This corresponds to the boot time of the internal processor and doesn t indicate any problem or failure in the audiocodec 5 3 Establishing a communication Several operating modes are available depending on the protocol used for the communication initiation 49 AEQ PHOENI X ALI O The simplest yet still very effective mode that we recommend for a start is IP mode RTP Raw wit
90. he procedure is described in detail in the following Application Note we recommend you to read it when you decide to use this working mode AEQ PHOENIX AUDIOCODECS APPLICATION NOTE 0 C Connecting AEQ Phoenix units through Internet Complex scenario configuration Through LOCAL network s DHCP not used manual NAT Making use of AEQ Proxy SIP The eight parameters to be configured in the dialog for this mode are ControlIPHOENIX Configuration PRG NAT traversal ALIO Select NAT mode MANUAL router configuration J Manual mode SIP local iP 192 168 1 86 ATP local iP 192 168 186 SIP local port 5066 ATP local port 5004 SIP public IP o 0 0 0 RTP public IP SIP public port 0 RTP public port c 1 SIP LOCAL IP read only parameter that tells you the IP of the IP interface of the unit as regards SIP so that the latter can in turn convey this to the router or firewall administrator when it is configured For instance 172 26 33 35 It can be set in order to adapt it to network necessities in menu Configuration gt Ethernet ControlPHOENIX u SE config Ethernet module 1 Enable DHCP IP Address me 168 1 Subnet mask 255 255 255 Gateway IP 192 Ee DNS Server 8 8 8 for changes to apply the equipment need to reboot Communication will be re es in aprox 15 sec 42 AEQ PHOENI X ALI O 2 SIP LOCAL PORT read o
91. his manual O E E 6 5 1 1 Ending an IP communication in RTP mode e To finalize the communication simply press the CALL button in the individual codec control window and then confirm As SmartRTP mode is active it won t be necessary to manually terminate the call at the remote end 6 5 2 Establishing an IP call in PROXY SIP mode e Ensure that the equipment is powered up and controlled by the software e Establish the appropriate audio configuration mixer e Check that there is incoming audio to the channel PROG or COORD that we are going to use to establish the communication the Tx indicator in the individual codec control ia NEE window in the general configuration screen and in the list view will change to green e Go to general configuration screen and configure INTERFACE as SIP Proxy based REGISTERING e Enter I F Setup and click on SIP Parameters INTERFACE ControlPHOENIX Configuration PRG SIP parameters ALIO Local uri Proxy SIP Proxy SIP Accounts Username phxalio_xxx displayname Phoenix Channel 1 Proxy Provider ES Authentication User phxalio_xxx AEQ Password aeq Manage Providers Realm sip aeq es 59 AEQ PHOENI X ALI O Check the SIP server configuration Proxy Provider Select one that is already configured from the list for example the defau
92. his Limited Guarantee or otherwise shall AEQ S A be liable for incidental special or consequential damages derived from the use or from the impossibility of using the product AEQ shall not be liable for loss of information in the disks or data support that have been altered or found to be inexact neither for any accidental damage caused by the user or other persons manipulating the product 13 AEQ PHOENI X ALI O APPENDIX A General characteristics of encoding modes OPUS OPUS is a completely open and very versatile coding algorithm Its performance is unrivaled for voice and audio transmission It was standarized by Internet Engine Engineering Task Force IETF as RFC 6716 and combines Skype s SILK codec technology with Xiph Org s CELT This algorithm allows for an excellent audio quality with high compression rate and very low delay Phoenix family audiocodecs feature 7 selected OPUS modes covering nearly every transmission need from voice to high quality stereo music with bitrates between 12 and 192 Kbps and audio bandwidth between 6 and 20 kHz The receiver can automatically adapt to the particular OPUS mode selected in the transmitting end G 711 ITU encoding standard for processing audio signals in the human voice frequency band through the compression of digital audio samples obtained at 8KHz and typically used in telephone systems Bandwidth 3 5 KHz For further information on this subject consult http www it
93. ial number of the product It will be equally necessary the previous and expressed conformity from the AEQ Technical Service for the shipment to AEQ of products for their repair or substitution in application of the present guarantee As a consequence returns of equipment that does not comply with these conditions will not be accepted 3 AEQ will at its own cost repair the faulty product once returned including the necessary labour to carry out such repair whenever the failure is caused by defects of the materials design or workmanship The repair will be carried out in any of the AEQ authorized Technical Service Centres This guarantee does not include the freight charges of the product to or from such Authorized Technical Service Centre 4 No Extension of the Guarantee Period for repaired product shall be applied Nor shall a Substituted Products in application of this Guarantee be subject to Guarantee Period Extension 5 The present guarantee will not be applicable in the following situations Improper use or Contrary use of the product as per the User or Instruction Manual violent manipulation exhibition to humidity or extreme thermal or environmental conditions or sudden changes of such conditions electrical discharges or lightning oxidation modifications or not authorized connections repairs or non authorized disassembly of the product spill of liquids or chemical products 6 Under no circumstances whether based upon t
94. iguration ALIO IA Autoanswer Autoanswer Number URI Leave blank for any Auto hang up if sync loss we after ch seconds Permanent call Apply audio profile to incoming calls o Autoanswer Automatic call answering for all incoming calls or only those corresponding to a predefined caller o Auto hang up Automatic hang up whenever audio packets are missed for a given time o Permanent call The device will do what s necessary to keep the call up even after a line drop or mains cut o Apply audio profile to incoming calls allows you to filter the SIP calls depending on the encoding profile of the receiver Return to the individual codec control window and click on CALL button corresponding to that channel showing then the call screen ControlPHOENIX a ALIO Call on PRG Call to gt phx55 192 168 1 55 D Channel interface SIP Proxy based Calls EL Equipment contacts ES d 192 168 1 68 5008 Channel 1 S 15 01 2010 09 38 27 00 02 35 192 168 1 68 5008 Channel 1 gt 15 01 2010 09 3359 00 00 07 Enter the URI of the remote unit either manually or E E getting it from the buttons By clicking here the last URI Calls the URIs in the Equipment Contacts book or the available IP addresses are shown respectively but only those with formats compatible with channel and communication type 64 AEQ PHOENI X ALI O PHOENIX
95. it e Private IP addresses both static and dynamic corresponding to connections in a local network with several workstations that access to the Internet through a router with NAT Those do not allow the use of URIs of the name IP address type because the IP address of the identifier is not public and is only valid in the section of the network to which it has been assigned it lacks a universal meaning In this case the use of an associated SIP server and a STUN server is imperative to get past the NAT Network Address Translation implemented in the router that acts as an interface between the private network and the public one See section NAT TRAVERSAL 4 3 B4 3 PHOENIX ALIO default SIP configuration To simplify operating the unit AEQ offers at no additional cost the services of 2 own SIP servers PHOENIX ALIO is supplied from the factory with both SIP servers preconfigured SYSTEM gt SIP PROVIDERS menu defined as AEQ and AEQ 2 with the following configuration PROXY SIP AEQ Host sip aeqg es PROXY SIP AEQ 2 Host sip2 aeq es PROXY SIP AEQ and AEQ 2 Port 5060 PROXY SIP AEQ and AEQ 2 Domain sip aeq es PROXY SIP AEQ and AEQ 2 Register Expires 60 min PHOENIX ALIO is supplied preconfigured with 2 users registered in both servers e PROXY SIP AEQ and AEQ 2 Authentication Users phxalio XXX y phxalio_XXY where Y X 1 e PROXY SIP AEQ and AEQ 2
96. le is connected and latched Check that the amber LED integrated in the RJ45 port is flashing Establish the desired audio configuration inputs routing mix level MIC4 LINE IN mode and outputs If the coordination channel license is activated select the channel we want to call from by means of the NAVI Ch button The corresponding legend will get illuminated between the vumeters PROG or COORD Press the SIP key until PROG COORD CHANNEL PROXY SIP MODE is displayed Press the MENU key and select COMMUNICATIONS Under SIP menu verify that a valid PROVIDER is selected a valid user is entered and the password is correct Also within COMMUNICATIONS go to the NAT menu and check that a NAT TRAVERSAL mode adequate for your connection kind has been selected Verify the status of the communications Check that the SIP key is steadily illuminated If it blinks there is some problem with the Ethernet connection or the access or registration on the Proxy server and the unit won t allow you to make any calls If this is the case check connectivity and configuration under the ETHERNET SIP and NAT menus carefully all of them accessibly with the MENU key Select a coding profile according to the desired quality and network capabilities by pressing the CODEC button Establish the desired automatic options Auto Answer auto hang up and permanent call by means of the AUTO button Press th
97. lgorithms but it is also compatible with other AEQ and third party audiocodecs as it also features AEQ LD Extend modes and the mandatory algorithms according to EBU TECH 3326 specification from EBU N ACIP work group When connecting to another AEQ codec users can take advantage of an exclusive set of tools that makes initating communications and control of the unit a simple task e The SmartRTP proprietary call initiation protocol that simplifies connection to compatible codecs e AEQ ControlPHOENIX remote control Software that allows for the remote operation and adjustment of the unit from your station ControlPhoenix allows you to control everything related to the call initiation process and also the adjustment of all audio parameters and the local audio routing of ALIO e HELP function that allows the journalist to use the system to request for assistance from the station when facing an unexpected situation By default PHOENIX ALIO offers a stereo or mono channel for the program signal with its corresponding return A second bidirectional mono or stereo channel for coordination or redundancy purposes can be activated by purchasing its license that activation can be accomplished from AEQ ControlPHOENIX application and is detailed in section 6 4 of the application user s manual PHOENIX ALIO is powered from mains Optionally it can be equipped with an UPS that is mains charged and can provide more than 120 minute
98. licking on this link an Internet browser will pop up showing Phoenix ALIO Web management window allowing among other things to update firmware and obtain real time IP traffic statistics when the channel s is are connected When the unit has no license for COORDINATION channel activation the screen will look slightly different as the control area for that channel will appear deactivated PHOENIX Y ALIO JE ii not connected DISABLED ALIOL WRI 1 alio 15 prog URI 2 alio 15 coor The left zone shows the general status of both communications channels PROGRAM PRG and COORDINATION CRD CONNECTED OK REGISTERING etc as well as the remote equipment s data IP address and port or name in case it is connected We can click in any of both areas in order to show a window that provides all the details of the channel we have clicked on 29 AEQ PHOENI X ALI O AEQ alio ssm INTERFACE Statys CONNECTED CONNECTED TO 192 168 1 66 5008 e INTERFACE indicates the operating mode of the channel RTP Raw DIRECT SIP or Proxy SIP e Coding indicates the coding algorithm or profile OPUS G711 G722 MPEG L2 lt SIP CODEC PROFILE gt This section also indicates binary rate 128 Kbps for example the sampling frequency 48KHz for example and the mode Mono Stereo Dual JStereo or MS Stereo e Status SIP registering status for IP connections using Proxy SIP mode REGISTERING REGISTRATION ERROR
99. lt AEQ or otherwise go to Manage Providers create a New provider and fill in the following fields name port address of the server either its IP or URL SIP interface and if necessary check the Register field In this case you also need to re write the Authentication data in the channel s SIP parameters so they match those in the new server as the ones by default are only for ControlPHOENIX New provider w n insert new provider values Description MY SIP PROVIDER i Host sip myowndomain com Port sen Domain sip myowndomain com Register Expires AEQ Note that in case you specify the SIP Server by its URL instead of its IP address the DNS Server must be correctly configured and reachable at Configuration gt Ethernet You can also select a SIP account from a previously created and stored list by means of the drop down menu Proxy SIP Accounts See section 5 1 7 1 of AEQ ControlPHOENIX user s manual Select the working mode to traverse NAT devices NAT Traversal that is more adequate for the network the unit is connected to NOTE It is recommended that you follow Application Notes 0 A or 0 C according to the type of equipment s connection At I F Setup fill in the Local media port where the unit expects to receive RTP audio traffic at If you enable Symmetric RTP mode the unit will send audio to the same por
100. makes the unit automatically accept the incoming calls On the contrary and if in manual answer mode incoming calls are signaled on the screen and the user must press the OR key to accept them or the ESC DEL key to reject them 3 7 2 2 Auto Hang Up Next the Auto Hang Up option is available in the AUTO menu This option makes the unit hang up a call whenever the incoming stream can t be synchronized either because it is not received in this case the call status will be CONNECTED NO DATA or because some incoming packets are damaged or lost or because the encoding algorithm is not the one expected In this case the call status will be CONNECTED NO SYNC In both cases the call will be disconnected or hung up once the set out time for synchronization has been completed timeout This is useful and avoids leaving a unit busy indefinitely or when it is needed to force re dialling from the other end in SIP mode 3 7 2 3 Permanent call This option configures the unit so that it automatically does what is necessary to maintain the active connection even after temporary line drops or power outages It is recommended that this option is activated in one of the communicating audiocodecs only except when using RTP with SmartRTP mode deactivated In this case this option must be enabled at both sides to guarantee the operational efficiency 3 7 3 CODEC key CODEC This key allows the sele
101. mately 15 seconds If you have any doubts please consult your IT network technician or directly contact the AEQ or authorized distributors technical support department 4 7 SNMP Configuration This unit can be remotely managed using SNMP Simple Network Management Protocol using one of the many client pieces of software available in the market even for free SNMP allows monitoring of the status of several pieces of equipment from very diverse manufacturers and natures as well as elaborating reports generate e mail alerts etc You can access the configuration menu in Configuration gt Network 46 AEQ PHOENI X ALI O controlPHOENIX y Network management aeq 03 Remote control SNMP SysLog DAMME Send traps toIP1 12 25 25 26 Protocol version MI Send traps to IP2 0 0 0 0 Protocol version MI Send traps to IP3 0 0 0 0 Protocol version VI Insert 0 0 0 0 to disable the traps sending AEQ PHOENIX Audiocodecs Mercury Venus Studio Stratos and ALIO can connect to up to 3 SNMP clients installed in remote PCs by simply configuring their IP addresses in the SNMP tab of previous menu Once one or more SNMP clients are connected and the corresponding MIB descriptive file has been loaded it can be downloaded from the equipment s Web Interface see section 7 8 of this manual the audiocodec will send accept different types of informations to from each client 1 Alarms Traps
102. meters of the connection o SAP Session Announcement Protocol for multicast type unidirectional links o SIP Session Initiation Protocol simulates the working system in traditional telephone networks e Transport defines the transport protocols over IP networks o RTP Real Time Transport Protocol over UDP and IPv4 o RTCP Real Time Control Transport Protocol for synchronization and active retrieval functions o IP ports defined 5004 5008 RTP for PROG COORD respectively and 5005 5009 RTCP While this appendix is not intended to be a reference document for all the relevant technical matters it should at least serve to give its readers an initial contact with these subjects that will ease the assimilation of the new working method over IP networks for the Phoenix ALIO user and as a result the use of this equipment The user interested in expanding his or her knowledge of some or all of these subjects is encouraged to turn to the extensive excellent technical material currently available regarding the IP realm and the technologies associated with it B1 Circuit switching versus packet switching The communications systems traditionally used in the broadcast environment for applications with portable codecs have been mostly telephone or ISDN networks that is circuit switching networks Phoenix ALIO on the other hand uses a packet switching network in its IP interface B1 1 Circuit switching In a circuit switching network
103. most of the parameters involved in this NAT TRAVERSAL menu and the importance that any modification has in the final operation of the unit we recommend that only highly qualified personnel in possession of all the technical documentation and manuals work on this NAT configuration menu For additional information see APPENDIX B5 The NAT traversal options of a codec are accessed by following this sequence from the involved individual codec control window Configuration gt General gt I F Setup gt NAT Traversal Next we will describe the operation without NAT and the other five modes supported by Phoenix ALIO 41 AEQ PHOENI X ALI O 4 3 1 Operation without NAT OFF there is no NAT The unit uses no mechanism to traverse devices with NAT This mode will be used only to operate in the local network all of the SIP participants are in the same local network including the Proxy SIP if we use it 4 3 2 Manual NAT MANUAL router configuration This mode will be used when the unit is connected to a local network with shared Internet access through a router that will work as NAT Network Address Translation In order to use this mode no DHCP must be used and you need to have access to router configuration and the knowledge to do it or to the Network Administrator that will give us some data to be configured in the unit and configure the router to open and redirect some IPs and ports port forwarding T
104. n 6 1 3 1 Coding selection at AEQ ControlIPHOENIX user s manual Although AEQ recommends the use of Fe OPUS coding algorithms for most uses several different modes are available to match almost any need a coding ut formats RTP raw dbclick to select Coding selection PRG S ao selection window can be accessed by Type Mode Samplerate BitRate Bits sample Law a S Ge OPUS VOICE 48 Khz 12 Kbps clicking on the Select codec button ares located inside the ENCODER area of cis EC SC G e D D OPUS MUSIC STEREO 48 Khz 128 Kbps the general configuration window where os MUSIC STEREO 48 khz 192 Kbps G722 MONO 16 khz 64 Kbps both OPUS modes and others provided 2 g t ck ee for compatibility can be found are STEREO eae Oe AEQ LD STEREO 32 Khz 256 Kbps AEQ LD STEREO 48 Khz 384 Kbps tc II H hz bps Note that the DECODER will be Fa MONO SS 512Kbps 16 g PCM MONO 32 khz 640 Kbps 20 automatically configured for the same MONO 32 khz 768 Kbps 24 PCM MONO 48 Khz 576 Kbps 12 coding algorithm and mode POM MONO a eru 20 PCM MONO 48 Khz 1152 Kbps 24 PCM STEREO 32 Khz 768 Kbps 12 PCM STEREO 32 Khz 1024 Kbps 16 PCM STEREO 32 Khz 1280 Kbps 20 PCM STEREO 32 khz 1536 Kbps 24 However when the interface is IP and configured in any of the SIP modes DIRECT IA in curt conn gt Lopterofl HH SIP or PROXY SIP E am particular coding algorithms gt MIO wont
105. nced button to specify the selected coding method and add replicas to the list The OPUS MUSIC STEREO 48 kHz 64 kbps mode will be used in this example configuring the contact to issue replicas to 3 different IP Port pairs we have used the New Replica button in order to do that 4 Interface Advanced Configuration Type Mode SampleRate BitRate DitefGample Law 5722 MONG 16 khz 64 kbps AEQ LO ON 16 khz 64 Kbps AEQ LO ON 32 khz 128 Kbps AEQ LO ON 48 khz 192 Kbps AEQ LO STEREO 16 khz 128 Kbps AEQ LO STEREO 32 khz 256 Kbps AEQ LO STEREO 45 khz 364 Kbps DChM ONG 32 khz 384 Kbps MOMA ar Khe 517 khns A A New replica Edit replica Delete replica IdReplica IP 1 172 26 33 44 2 172 26 33 22 Detail of the creation o RTP replicas within a General agenda contact 39 AEQ PHOENI X ALI O We need to select this contact that includes replicas from the call window when making a call ControlPHOENIX CO a ALIO E ControlPHOOMIx Call on PRG F x p Call to hee Contact L t certect Deiete contact Lod Saws Chose E gt AEE Tri SS Cenegal contacts Gereral Filtered by RTP Intertace g d Genra ail Descrip b iert sce Bees l Bee 2 Channel interface RTP raw nacht EZS E am Calls 2 Equipment Contacts EEN P 2 Replicas in Contact 192 168 1 68 5008 Channel 1 2 gt 15 01 2010 12 14 44 00 04 46 192 168 1 68 5008 Channel 1 gt 15 01 2010 12 12 29 00 58 59 e Fr
106. ncoder and the rest of keys and encoders Details will be provided later on in this chapter The display also provides detailed information about active calls if any To save power and if the unit is idle the intensity of the display is dimmed after a while normal brightness will be immediately recovered when any control is touched AEQ PHOENI X ALI O 3 2 Navigation Channel encoder NAVI Ch This rotary encoder allows the user to browse through the different menus of the OLED multifunction screen Turning it changes the selection among the options presented moving the highlighting up and down When the encoder is turned clockwise the option selected is moved down and turning it anti clockwise the option selected is moved up selection at that moment Pressing its button is equivalent to ENTER validating the highlighted NAVI Ch This rotary encoder has a second use If the the second optional channel coordination COOR is activated and no menu is being presented on the display pressing the encoder will make the user interface change between PGM and COORD Tha modification of any of the parameters pertaining to the selected channel does not affect the selected operation of the other channel if we are modifying the configuration for program PROG the status for coordination COORD will remain the same with its communication operating modes etc unaltered The active channel is clearly indicated by th
107. ng on the microphone model and its specifications the negative of observing this procedure may result in very high level and high pitched noise that could be routed to the units communication buses and the headphone outputs The corresponding input encoder can be used to navigate within the menu or alternatively the NAVI Ch rotary encoder can also be used The BACK option in each menu or the ESC DEL key can be used to cancel and go back MICI GAIH 3206 EQ off See off MICi eeng IC Lee HE GAIN 32d6 ES off EH off FHAHTOM of F PHOTO EACK EACK erste ae MIci Bde AEQ PHOENI X ALI O The EQ indication in the main screen input name highlighted will become active whenever a value different to O dB is selected for either the bass or treble controls as previously explained The MIC4 LINE input menu is slightly different as there is an additional option to select between MIC4 and LINE IN MIC4 LINE INPUT LINE EW ott BACE The fourth input is a line input in this example Note that the Phantom option becomes unavailable until the input is switched to MIC4 NOTE When browsing an input menu the user can quickly change to configure another input by simply pressing the corresponding encoder button From now on the modifications made correspond to the newly selected input 3 6 Outputs control F The unit has 3 stereo outputs 2 for headphones able to
108. ng that the Tx and Rx audio presence indicators change to green 66 AEQ PHOENI X ALI O 7 CONTROL TERMINAL OVER WEB BROWSER PHOENIX ALIO audiocodec includes a WebServer that enables you to execute numerous functions remotely over the Ethernet interface included in the back panel of the unit by means of a standard web browser compatibility is guaranteed with Internet Explorer running on Microsoft Windows operating system 7 1 Upgrading system firmware PHOENIX ALIO is supplied from factory with the latest firmware versions available However firmware versions with new features may be released in the future making it necessary to upgrade the equipment to be able to make use of these new functionalities Because the upgrading process must be handled with caution we recommend having it done by an authorized distributor or under the instructions of the AEQ Technical Assistance Service If questions or problems arise please consult via electronic mail sat aeq es IMPORTANT NOTE If the CPU of the equipment is upgraded configuration of the unit and in particular IP configuration won t be modified unless expressly stated by AEQ SAT In tat chase the user should take note of all important codec parameters before upgrading in order to reconfigure them afterwards The entire PHOENIX ALIO firmware versions upgrading process is done through the IP interface of the unit with the aid of a standard web browser compatibility is
109. ng the list entry CUSTOM If this is selected the user can manually set the parameters by entering its description its IP either host name or IP address domain and port The same procedures explained to enter IP addresses in the Ethernet section are applicable here PROVIDER option won t be available whenever PROXY OFF Last the unit URI username can be specified within the SIP ACCOUNT option When PROXY SIP is activated some additional registration parameters will be required here password presented name and subscription expiration timeout NAT The kind of strategy to traverse routers performing NAT see appendix is specified here This option will only appear when the active channel has been configured in SIP mode with the corresponding key see 3 7 1 HAT HAT MODE auto STUH BACK 26 AEQ PHOENI X ALI O Phoenix ALIO provides for SIP connections a total of six different modes when traversing NAT devices routers firewalls Each of these modes is the most adequate for a certain scenario For example when the involved audiocodecs are within the same local network the strategy wont be the same when working through an Internet connection with a dedicated router multihost with NAT etc One or other mode is more convenient as a function of these circumstances NO NAT AUTO1 AUTO2 AUTO3 AUTO4 or MANUAL In order to specify one select the NAT MODE option and choose one of the available modes
110. ng to dial from both ends and without worrying about the coding algorithms matching as the calling end will provide all the necessary signaling to its counterpart so it knows what IP address and port to send the return audio to This functionality is in some sense similar to the one provided by DIRECT SIP but without renouncing to RTP inherent simplicity all other modes are based on RTP indeed and without the need for additional special control ports In order to activate SmartRTP press the AUTO key and enable this option 6 5 1 Establishing an IP communication in RTP mode using SmartRTP e Ensure that the equipment is powered up and controlled by the software e Establish the appropriate audio configuration mixer e Check that there is incoming audio to the channel PROG or COORD that we are going to use to establish the communication the Tx indicator in the individual codec control window in the general configuration screen and in the list view will change to green e Go to general configuration screen and configure INTERFACE as RTP raw LF Setup e Enter I F Setup and select the Local media port local IP port through which the RTP audio is received Ensure that the remote unit when calling sends audio to that port see section 4 2 1 Local media port 3 Adaptive Adaptive buffer max Mm OFived 57 AEQ PHOENI X ALI O The same screen allows you to configure the type
111. nly parameter that tells you the port of the IP interface of the unit used for SIP signaling so that the latter can in turn convey this to the router or firewall administrator when it is configured Before checking the value of this parameter you should have configured previously whether you want to work with Proxy or not and restart the unit 3 SIP PUBLIC IP parameter that will tell the unit which public IP will correspond to it so that it can include the said IP in its SIP messages The router or firewall administrator must tell you the value of this parameter For instance 212 170 180 177 4 SIP PUBLIC PORT parameter that will tell Phoenix which public port it will have corresponding to its local SIP port The router or firewall administrator must tell you the value of this parameter in order to make the required port forwarding For instance 8001 5 RTP LOCAL IP read only parameter that tells you the IP of the IP interface of the unit as regards RTP so that it can in turn convey this to the router or firewall administrator when it is configured You will usually configure the same network interface as for SIP so it will be the one configured in point number 1 for instance 172 26 33 35 6 RTP LOCAL PORT read only parameter that tells you the port of the IP interface of the unit as regards RTP so that the latter can in turn convey this to the router or firewall administrator when it is configured Usually the shown port is 5004
112. nnection with the STUN server is normally made through port 3478 by means of UDP The STUN server can then provide the client with an alternate IP and communication port For complete information on the STUN protocol we recommend consulting http tools ietf org html rfc3489 AEQ always has a PHOENIX unit available for test at phoenixMaster sip aeq es URI and its 2 SIP servers are also available at sip aeq es and sip2 aeq es and with warranty that both work according to the official standard 85 AEQ PHOENI X ALI O APPENDIX C Ports used by PHOENIX equipment When Phoenix unit is installed in a private IP network and you want to establish communication with other units through that network router firewall gateway three indications related to the ports used by the unit must be taken into account 1 Output permissions in router firewall Phoenix unit will send packets to different servers and or other units each one will use a different port Therefore firewall will have to allow that packets from Phoenix unit IP are sent towards the following ports Protocol Port number Usage DNS domain name server protocol UDP 1462 SNMP Traps pot lt i lt OO E ees 5010 RTP protocol audio packets going towards the remote may change unit ports depend on the remote unit network not Phoenix one eee 3479 STUN protocol a ee ro for getting the public IP of the nee poo a ee ro 5060 5062 SIP protocol S
113. om a single sender broadcast in which the recipients are all the stations in the network and anycast transmission to a single recipient any unspecified recipient The unicast method is the one currently being used on the Internet and is applied for both live and on demand transmissions The multicast method can only be used in corporate environments despite some isolated efforts to introduce it on the Internet and is applied only for live transmissions 19 AEQ PHOENI X ALI O SA Graphical comparison Unicast vs Multicast The effect that unicast transmission has on network resources is accumulative consumption Each user who connects to a multimedia transmission consumes as many kilobits per second as the content encoding will permit B3 RTP protocol RTP are the initials of Real time Transport Protocol It is a transport level protocol used for the transmission of information in real time as occurs with audio and video Normally it is paired with RTCP RTP Control Protocol and is located on UDP The IP ports defined for its use are 5004 RTP and 5005 RTCP for PROG and 5008 5009 for COORD The functions of the RTP RTCP protocol are e Management of the reception buffer in order to minimize the jitter effect introduced by the network e Recovery of the reference clock based on information inserted by the transmitting equipment e Test tools to permit the user to verify the bandwidth the delay and estimated jitt
114. ome Phone PF URI 192 1en Leo Once the available contacts are displayed select the one we are interested in replica3 in this example by double clicking its name and we will be returned to the call window At this moment we are able to check that the specified main IP address and port fields fill the Call to field and we can also check that the replicas are going to be loaded into the unit by clicking on the Replicas in Contact before actually making the call ControlPHOENIX A ALIO t Call on PRG gt Call to 92 168 1 68 5008 D Channel interface RTP raw a Calls 2 Equipment Contacts EES Ip 2 Replicas in Contact Num Destination Enabled FEC if FEC mode is 1 1 192 168 1 69 5004 NO D 2 192 168 1 69 5008 NO Once we click on the green Call button the unit will start emitting the audio streams to the main address and to the specified replica addresses ALIO will stop sending replicas as soon as the call is hung The list will be erased from the unit and the only way to send them again is to call selecting the same contact However if the Permanent Call mode is activated and there is a mains cut for example the unit WILL send all configured replicas when rebooting NOTE 1 When audio is transmitted to several destinations it can be received from only one of them or none In order to establish which of the units sends the audio back and only in RTP
115. onnected to the CPU and consists in an OLED display where the control and configuration menus are displayed as well as a set of keys and indicators associated to the operation of the unit such as the alphanumeric keyboard rotary encoders and four high resolution LED vumeters The CPU is a high performance and low power ARM processor in charge of several tasks such as the user interface configuration of the other programmable elements DSP FPGA audio processor preamplifiers etc and the management of IP communications etc The DSP FIXED POINT DSP is a high performance fixed point processor that carries out the encoding and decoding of up to two stereo channels using different compression algorithms and as later described in this manual The audio matrix AUDIO MATRIX amp MIXER is implemented using a new generation low power FPGA with 6 inputs 2 stereo receiving buses 6 outputs 2 stereo transmitting buses The FPGA can perform any crosspoints combination with great dynamic range and is controlled by the CPU It also relies on a specialized co processor AUDIO PROCESSOR that provides individual low and high frequency adjustment for each input AEQ PHOENI X ALI O The network interface NET I F is an Ethernet 10 100 Mbps interface that allows for both audio transmission reception and remote control of the unit through a single port The functionality of this port can also be for the connection of standard equipment to
116. ork you need to change their IP address one by one as you add them in order to avoid conflicts in the network as they will also have the same default IP addresses Go to Configuration gt Ethernet to access the dialog that allows you to change the IP parameters of the unit 6 4 Audio Chapter 2 in this manual describes the physical connections present in the back and side panels of the audiocodec in detail but in a nutshell this is the simplified procedure e Connect the required microphones to inputs MIC1 to MIC4 at the back panel e Select an adequate gain usually around 40 dB using each input s configuration menu accessible by simply pressing the corresponding encoder button MIC1 MIC4 LINE e Activate Phantom supply where necessary check the microphone manual e If required connect the line input source to the XLR connectors at the right side of the unit labeled as IN L amp R e Connect one or two headphones to the HP1 and or HP2 outputs at the right side of the unit e f required connect the line output to the XLR connectors at the right side of the unit labeled as OUT L amp R 6 5 Establishing an IP communication As explained in previous chapters several operating modes are available depending on the protocol used for the communication initiation In order to ease the task AEQ has developed the proprietary SmartRTP protocol that allows for the establishment of a communication without havi
117. ormation please consult section 7 7 of this manual and section 6 5 1 of AEQ ControlPHOENIX application manual 48 AEQ PHOENI X ALI O 5 QUICK USER S GUIDE LOCAL OPERATION In order to deeply know the operation of Phoenix ALIO unit it is strongly recommended that the previous chapters are thoroughly read In this chapter the basic actions to manually operate the unit are described If more detail is needed please check the information provided in the previous chapters 5 1 Connecting the unit 5 1 1 Power supply Power supply to the unit is provided by the provided AC DC adapter unit or by means of a homologated UPS In any case connection to the unit is made by means of the special latching connector at the back as described in chapter 2 2 3 of this manual 5 1 2 Audio Chapter 2 in this manual describes the physical connections present in the back and side panels of the audiocodec in detail but in a nutshell this is the simplified procedure e Connect the required microphones to inputs MIC1 to MIC4 at the back panel e Select an adequate gain usually around 40dB using each input s configuration menu accessible by simply pressing the corresponding encoder button MIC1 MIC4 LINE e Activate Phantom supply where necessary check the microphone manual e If required connect the line input source to the XLR connectors at the right side of the unit labeled as IN L amp R e Connect one or two headphones to t
118. ously based on the Link Profiles SIP Codec Profiles defined in each of the devices at the two ends of the connection circuit B4 2 Possible work scenarios Depending on the type of network to which the PHOENIX ALIO is connected the codec will have one or another type of IP address available to it e Static public IP addresses offer the ideal situation since they guarantee that the IP interface of the codec will always be assigned to a fixed address regardless of whether it is turned off and then powered up again and directly accessible to the rest of the network users Phoenix ALIO operates perfectly with an associated SIP server and equipment identifiers of the name domain type PROXY SIP and even without an associated SIP server with a URI of the name IP address type DIRECT SIP if the device on the opposite end of the communication circuit also has an IP address of the same type This situation corresponds to use an Internet access by means of a single workstation router just one piece of equipment connected and to hire a fixed IP e Dynamic public IP addresses corresponding to use an Internet access by means of a single workstation router and a dynamic IP the most usual Allows the use of URIs of the name domain PROXY SIP or name IP address DIRECT SIP type but it is advisable always to work with an associated SIP server PROXY SIP since the IP address assigned to the equipment may change each time the user powers up the un
119. pdated to firmware version 5 20 or above 1 6 Compatibility with third party codecs PHOENIX ALIO is a portable IP audiocodec compatible with EBU TECH 3326 technical specification from EBU N ACIP workgroup This technical specification was developed to guarantee compatibility between equipment from different manufacturers in applications for professional quality audio contribution over IP networks Therefore it is possible to connect PHOENIX ALIO with any codec from other manufacturer over IP provided that this unit has been developed according to N ACIP please check third party codecs technical specifications AEQ PHOENI X ALI O 2 PHYSICAL DESCRIPTION OF THE UNIT Before anything else it is necessary to become familiar with the connectors and other elements present in the back right and front panels of the unit in order to understand the wiring and installation required for the PHOENIX ALIO 2 1 Description of the right panel and connections 2 1 1 Headphone 1 and 2 outputs HP1 y HP2 a Ys Headphone Jack Unbalanced connection Common Shield Right Left a 14 Jack pinout 2 1 2 Line inputs LINE IN XLR 3 female connector Balanced connection Connectors as seen from the soldered side L input Female R input Female XLR 3 pinout Pin 1 gt Ground Pin 2 gt Input Pin 3 gt Input AEQ PHOENI X ALI O SA 2 1 3 Line outputs LINE OUT c XLR 3
120. phones and a stereo line output appear PROGRAM and COORDINATION send and receive buses as well as CUE arrive to this block in order to be able to output or monitor them The OUTPUT MIXER block allows for the configuration of the assignment of buses to outputs level control and send receive balance adjustment for each one The amplifier symbols at the inputs are filled with the color corresponding to the bus they are being routed to in order to ease a quick identification of the routing at a glance Each output on the other hand is colored with the color code of the bus that is being routed to it NOTE The Config Mix button opens the complete mixer that controls both the inputs and the outputs just the same as the MIX button located at the right side of the individual codec window as you can see in section 4 1 2 Other available options within the CONFIG menu are Contacts call book management Ethernet IP configuration of the device Miscellaneous various adjustments and Network auxiliary network functions configuration The details of these menus are described in chapter 6 of AEQ ControlIPHOENIX users manual The Miscellaneous option allows you to activate the COORDINATION channel if the corresponding license has been purchased Just click on the CONFIG button again to close this menu 4 1 2 Mixer control window Also at the right side the MIX button opens
121. port to IP address port type Obviously there is no advanced signaling protocol in this scenario and you will need to established set parameters and disconnect communication from both ends Audio encoding must be the same and explicitly specified in both ends of the communication In order to avoid that hassle making the calling hanging up and coding selection tasks easier as it will be necessary to do it from one end only SmartRTP can be activated in both involved audiocodecs provided that they are AEQ Phoenix compatible with this mode If the operating mode required for a contact is RTP raw the only valid equipment identifier is lt IP_address gt lt destination port gt for example 172 26 33 28 5008 The specified destination port must match the Local port set up for the remote equipment That is in order to make a RTP call we must know the IP address and local port of the remote unit even if SmartRTP is used When you create a Call Book these fields describing a contact can be modified in the Call Book that can be accessed from a codec individual control window through the Contacts option in Configuration see section 5 1 7 of AEQ ControlPHOENIX user s manual Genera ES Ethernet network You can access the IP configuration submenu for RTP Raw mode by clicking on I F Setup button and that it is explained in section 6 1 4 3 of AEQ ControlPHOENIX user s man
122. rom by means of the NAVI Ch button The corresponding legend will get illuminated between the vumeters PROG or COORD e Press the SIP key until PROG COORD CHANNEL DIRECT SIP MODE is displayed Press the MENU key select COMMUNICATIONS and verify that the involved channel has been assigned a correct URI name e Also within COMMUNICATIONS go to the NAT menu and check that a NAT TRAVERSAL mode adequate for your connection kind has been selected e Verify the status of the communications Check that the SIP key is steadily illuminated If it blinks there is some problem with the Ethernet connection and the unit won t allow 53 AEQ PHOENI X ALI O you to make any calls If this is the case check connectivity and configuration under the ETHERNET and NAT menus carefully all of them accessibly with the MENU key Select a coding profile according to the desired quality and network capabilities by pressing the CODEC button Establish the desired automatic options Auto Answer auto hang up and permanent call by means of be AUTO key Press the OKT button The numeric keyboard will become illuminated in a color that depends on the active channel red PROGRAM or green COORDINATION Using the NAVI Ch encoder select whether you want to call from an agenda contact CONTACT LIST or manually dial an URI If manual mode is selected go to the URI line and fill the destination name as it
123. s is displayed o CONNECTING The ORT key will blink during this time o CONNECTED NO SYNC NO DATA When the call has been successfully established data is received but synchronization to it is not possible or no data at all is received respectively The OK key will remain steadily illuminated If status is NO SYNC or NO DATA and auto hang up option is enabled the call will be rejected after the defined time and the OKT key illumination will turn off Once connected with the remote end verify that the vumeters in Phoenix ALIO front panel show the presence of send and received audio and adjust levels as necessary 5 3 3 Establishing a DIRECT SIP call This mode is simply a variation of PROXY SIP using the same signaling and call establishment protocol but without the aid of an external server so that the destination IP needs to be known Note that since SmartRTP mode is offered in Phoenix audiocodecs the DIRECT SIP mode doesn t offer obvious functional advantages so its use will be typically relegated to cases where operation against third party or older units is required Check that the unit is ON Check that the RJ45 cable is connected and latched Check that the amber LED integrated in the RJ45 port is flashing Establish the desired audio configuration inputs routing gains mix levels MIC4 LINE IN mode and outputs e If the coordination channel license is activated select the channel we want to call f
124. s of autonomy and can also feed a 3G 4G router modem set and even charge mobile devices There is a quick start guide available in chapter 5 of this document However it is strongly recommended to carefully read this manual and the AEQ ControlIPHOENIX user s guide before using the unit 1 2 Technical characteristics 4 female XLR 3 microphone inputs Low noise preamplifier and switchable Phantom power supply 2 KQ input impedance 2 female XLR 3 line inputs with 9 KQ impedance OdBu nominal level max 20 dBu 2male XLR 3 outputs Output impedance lt 100 Q nominal level OdBu max 20 dBu 2 W Jack stereo headphone outputs with volume control and TX RX balance adjustment from front panel AEQ PHOENI X ALI O Communications interface IP Ethernet port interface RJ45 connector Two independent links can be established when an optional 270 channel is purchased using the same interface Satellite an external satellite interface can be connected to the IP interface 3G 4G telephony a 3G or 4G modem can be connected to the IP interface Wireless data links a wireless bridge WiMax or WiFi antenna can be connected to the IP interface Other features Front user panel with keyboard and rotary encoders OLED graphic display 2 stereo LED VU meters Operating temperature range 10 to 45 C 14 to 114 F Dimensions 242 x 210 x 60 mm 9 5 x 8 3 x 2 4 inches Power supply 12V DC
125. s the substitution of MIC2 input by a 2 kHz tone at 20 dBV level adjustable between inf and 18 dBV by means of the MIC encoder The mentioned tone will be sent to the bus where MIC2 is routed and the microphone will remain muted if it is connected NOTE All these test options will become automatically deactivated when leaving the TEST menu to avoid that they are accidentally left active BUZZER The unit buzzer signals incoming calls and can be activated or deactivated through this function it will sound once as a test whenever its status is changed from OFF to ON in this menu DEFAULT SETTINGS The systems factory settings can be restored at any time through this option However the units IP configuration will remain unchanged avoiding the loss of connectivity and remote control REBOOT The unit can be reset or rebooted in the event of unexpected or incorrect performace This is also useful for instance after performing a firmware upgrade The FW is installed but will not be applied by the unit until the REEBOOT option is selected This way a new FW version can be installed while the unit is being used but will not be adopted or have any effect until the live transmisi n is finished and the REBOOT option is selected 28 AEQ PHOENI X ALI O AEQ 4 CONFIGURATION AND OPERATION FROM REMOTE CONTROL SOFTWARE Configuration and operation of Phoenix ALIO can be carried out remotely by means of the application AEQ ControliPHOE
126. session of the required network information 1 2 3 From MENU gt ETHERNET check the IP address associated with the Ethernet interface Power down the PHOENIX ALIO Connect PHOENIX ALIO to the PC from which you are going to perform the upgrading process using a crossed cable Power up the PHOENIX ALIO Open the Internet Explorer web browser and in the address bar enter HTTP lt IP address obtained in point 1 gt Press ENTER and the main screen will be displayed Selecting the MAINTENANCE option will enable you to modify the MAC address of the Ethernet interface of the unit 68 AEQ PHOENI X ALI O AEQ PHOENIX Y ALIO C D 192 168 1 88 index htm Portable IP Audio Codec MENU MAINTENANCE UPGRADE STATUS ETHERNET MAC ADDRESS SE ut CR MACrar Acton MAINTENANCE 4 002101 012345 Apply SNMP MAC change screen detail Modify the value in the MAC field associated with the desired Ethernet interface Press the Apply button In the Internet browser go to the MAINTENANCE section and ensure that the MAC address is now the correct one 9 Power the unit down oS 7 3 Technical Assistance Service and on line manuals Clicking on the Support tab in the upper part of the screen will take you to AEQ website where you will find all the information you need to directly contact the AEQ Technical Assistance Service as well as all the technical information and manuals regarding the unit 7 4 S
127. t where it is receiving it from This is sometimes useful to overcome NAT routers Local media port Adaptive butter max ms The same screen allows you to configure the type and size of the receiving buffer and FEC parameters as a function of the IP network quality so we have the shortest delay while audio cuts are minimized or eliminated in poor quality networks see paragraph 4 4 of this manual in order to select the optimal buffer configuration depending on your application Adaptive Fixed Return to the general configuration screen check that the e copntrp selected encoding profile in the green ENCODER area i corresponding to that channel PROG or COORD is Coding correct or otherwise click on Select codec to change it There are several pre defined profiles containing several out particular algorithms each one with preference ordering They can be edited and more profiles can be added The called unit will accept the call using the first coding algorithm that it supports from the list independently of the profile selected in that unit at that time Advanced Decide whether you will use the advanced automatic connection options or not o Autoanswer Automatic call answering for all incoming calls or only those corresponding to a predefined caller o Auto hang up Automatic hang up whenever audio packets are missed for a given time o Permanent call The device will do what s necessary to ke
128. t to Phoenix Mobile However it becomes an easy and effective way to operate with the aid of SmartRTP 4 2 2 PROXY SIP This type of connection is selected when the Phoenix ALIO is used working together with an external SIP server that will provide connection with remote unit through any network even Internet without knowing its IP address Both units local and remote must be registered in SIP server which function is to maintain a database with the registered codecs storing their connection parameters IP address audio ports in order to ease the task of making calls between them even when connected to different networks In order to make a call in Proxy SIP mode you must take into account that for the URI or SIP identifier of the equipment in question you can use any of the following syntaxes o lt unit_names for instance phxalio_231 o phoenixMaster o lt unit_name gt c lt realm_SIP_ servers for instance ohxalio_231 sip aeq es or phoenixMaster sip aeq es o lt unit_name gt lt SIP_server _IP gt for instance phxalio_231 232 168 1 2 or phoenixMaster 232 168 1 2 where 232 168 1 2 is the AEQ s SIP server sip aeq es o lt unit_name gt lt SIP_ server IP gt lt Port gt when the SIP port of SIP server is not the 5060 SIP Standard port the identifier must include the used port for instance phxalio_231 sip aeq es 5061 When you create a Call Book these fields descri
129. tages e Delay in initiating communication A time interval is required to make the connection which entails a delay in the transmission of the information e Blockage of resources No use is made of the circuit during the moments when there is no transmission between the parties Bandwidth is wasted while the parties are not communicating with each other e The circuit is fixed The communication route is not readjusted it is not adapted at each opportunity to the least costly path between the nodes Once the circuit has been established no use is made of the alternative less expensive pathways that may become available during the session e Poor fault tolerance If an intermediate node fails the entire circuit crashes The connections then have to be re established from zero B1 2 Packet switching The sender divides the message to be sent into an arbitrary number of packets of the same size to which a header and the originating and destination addresses are added as well as control data that will then be transmitted through different communication media between temporary nodes until they reach their destination This switching method is the one that is used in today s IP networks It has emerged to optimize transmission capacity through existing lines The temporary nodes store the packets in queues in their memories which need not be very large B1 2 1 Switching modes e Virtual circuit Each packet is routed through the same
130. ters in Phoenix ALIO front panel show the presence of send and received audio and adjust levels as necessary 52 AEQ PHOENI X ALI O 5 3 2 1 Ending a PROXY SIP call e In order to finish the communication just press the ESC DEL key for a longer time making sure that the currently selected channel is the one we want to cut The ESC DEL key will blink red during disconnection and the display will show the DISCONNECTING status Both will disappear only when the call has been completely disconnected 5 3 2 2 Receiving and accepting an IP call in PROXY SIP mode If the unit interface is correctly configured and the Autoanswer mode is not active AUTO gt AUTOANSWER OFF when a call is received e As opposed to SmartRTP mode incoming SIP calls ARE signaled and unless the Autoanswer option is enabled the user can decide whether to accept or reject the call by means of the OK ESC DEL keys The unit will emit an acoustic signal It can however be disabled under the MENU gt MAINTENANCE gt BUZZER menu The OK key will simultaneously blink to warn the user e Information about the caller will appear in the OLED screen indicating the channel PROGRAM or COORDINATION where the call is coming to e The user can accept the call by pressing the OK key or reject it by pressing the ESC DEL key assuming that Autoanswer option is not enabled e lf the call is accepted call statu
131. the switching equipment must establish a physical path between the communication media prior to the connection between users This path remains active during the communication between the users and is cleared down or released when the communication ends Example Switched telephone network Its operation passes through the following stages request establishment file transfer and connection cleardown B1 1 1 Advantages e The transmission is made in real time e Dedicated resources The nodes that are involved in the communication use the established circuit exclusively as long as the session lasts 76 AEQ PHOENI X ALIO e Once the circuit has been established the parties can communicate with each other at the highest speed that the medium allows without having to share the bandwidth nor the use time e The circuit is fixed Because a physical circuit is specifically dedicated to the communication session in question once the circuit is established there are no losses of time for calculation and decision making regarding routing through the intermediate nodes Each intermediate node has a single route for the incoming and outgoing packets that belong to a specific session which means it is impossible for the packets to be disordered e Simplicity in the management of intermediate nodes Once the physical circuit has been established no further decisions need to be made to route the data from origin to destination B1 1 2 Disadvan
132. the vumeters in Phoenix ALIO front panel show the presence of send and received audio and adjust levels as necessary 55 AEQ PHOENI X ALI O SA In order to gain a complete knowledge of the operation of Phoenix ALIO we recommend reading the previous chapters and AEQ ControlPHOENIX user s manual carefully The paragraphs below describe the basic actions you will need to take for remote operation of the equipment by means of AEQ ControlPHOENIX application 6 QUICK START GUIDE REMOTE CONTROL 6 1 Equipment connections 6 1 1 Power supply Power supply to the unit is provided by the provided AC DC adapter unit or by means of a homologated UPS In any case connection to the unit is made by means of the special latching connector at the back as described in chapter 2 2 3 of this manual 6 1 2 Communications interface Connect an Ethernet cable CAT5 or better finished in an RJ45 10 100 BT to the LAN connector provided at the unit s back panel The selected cable must be straight when the connection is made from the unit to a communications device switch router For more information about the pinout of this port please check section 2 2 2 of this manual 6 2 Turning the unit on e Once the unit is connected to the power supply through its adapter the OLED screen will turn on after around 15 seconds showing AEQ logo and the audiocodec model name e Check that audio routing and levels are correct if they
133. tion and IP interface configuration I F setup button ControlPHOENIX Phoenix Alio General configuration CONNECTED To not connected ALIO 192 168 1 88 i PROGRAM INPUTS OUTPUTS d a L2 Gees eee STEREO bs INTERFACE RTP raw v 0K If the unit doesn t have a valid COORDINATION channel activation license the appearance will be slightly different as the control zones for that channel will appear deactivated ControlPHOENIX Phoenix Alio General configuration ALIO1 192 168 1 83 ENCODER DECODER Coding Boding INTERFACE The general view offers a graphic display of the audio flow inside the unit The equipments inputs are shown at the left entering the input mixer that can be open by clicking on the Config Mix button see paragraph 4 1 2 31 AEQ PHOENI X ALI O This mixer outputs three buses PROGRAM send depicted in red COORDINATION send depicted in green and CUE prelisten represented in blue The first bus is sent to the PROGRAM block where several aspects related to communications communication type ports etc coding algorithms etc can be configured The second bus is sent to the COORDINATION channel that if licensed allows the same configuration as PROGRAM The CUE bus is routed directly to the outputs block as explained below The output block is presented at the right where two stereo head
134. u int rec T REC G 711 e G 722 ITU encoding standard based on ADPCM algorithms for processing audio signals in the human voice frequency band through the compression of digital audio samples obtained at 16KHz for greater audio quality and clarity This is the internationally accepted mode for two way communication because of its low delay which is why it is the most used standard in commentator and sports broadcasting applications Bandwidth 7 KHz For further information on this subject consult http www itu int rec T REC G 722 e MPEG LAYER II Well known widely accepted encoding mode that is used when the delay is not important since MPEG modes always have a greater delay than G 722 modes There are 64kbps encoding modes with sampling rates of 48 32 or 24KHz and 128kbps encoding modes with sampling rates of 32 and 48KHz Bandwidth 10KHz to 15KHz For further information on this subject consult ISO IEC 11172 3 and ISO IEC 13818 3 AEQ LD AEQ proprietary mode based on the previous AEQ LD Extend mode that combines the low delay offered by G 722 with the greater bandwidth of the MPEG modes optimizing these two aspects PCM 12 16 20 24 bits Linear audio without any compression process For further information on this subject consult http www digitalpreservation gov formats fdd fdd000016 shtml Other different encoding modes can be taken into account according to specific needs of each client 74 AEQ PHO
135. ual ControlPHOENIX Configuration PRG ALIO Multiple unicast FEC mode OFF Local media port 229 EI See QO Adaptive Fixed buffer length 100 ms Fixed Symmetric RTP 34 AEQ PHOENI X ALI O The parameters to be configured are e FEC mode this option allows you to configure whether FEC Forward Error Correction is used or not there is a trade off for a bigger binary rate See section 4 4 e Local media port this option allows you to configure the value of the IP port selected to transmit audio at origin over IP Minimum value 1 024 Maximum value 65 534 Default values 5004 PROG and 5008 COORD e Adaptive Fixed and Adaptive buffer max Fixed buffer length this option allows you to configure the type and maximum size of reception buffer See section 4 4 e Symmetric RTP when SmartRTP mode is not activated this advanced option at least allows the user to force the local unit to send audio to the same IP and port from which it is receiving audio The destination port specified when making the call will be ignored when we receive packets from the remote equipment This option will allow you to connect to an audiocodec with unknown IP and or port because its behind a router with NAT for instance Please notice that RTP Point to Point is a complex configuration mode suitable for permanent connections that some equipment may not support Specifically it can t be used to connec
136. unication with each other through the Internet working with no Proxy DIRECT SIP mode or using the Proxy SIP provided by AEQ sip aeq es PROXY SIP mode The two configuration parameters available on screen for this mode are 1 STUN SERVER parameter that tells the unit the STUN server that will be used On the Internet there is multitude of public STUN servers By default the IP address of stun sipgate net server is configured 217 10 68 152 2 STUN PORT parameter that tells the unit the STUN server port assigned by the administrator By default 3478 43 AEQ PHOENI X ALI O NOTE in this mode the Phoenix ALIO behaves in the exact same way as the Phoenix Mobile unit when it is using a STUN server 4 3 6 AUTO 4 audio over internet This mode is equivalent to AUTOS but it will be used the SIP server is not the one provided by AEQ and there are problems with AUTO3 mode The configuration parameters are the same as for AUTO3 STUN server specification 4 4 FEC modes and reception buffer configuration FEC error correction mode Error correction is performed by sending redundant information that allows the receiver to recompose the lost data in case of not perfect transmissions Forward error correction always generates a higher binary rate and this in turn can generate more and more losses in very narrow transmission channels as well as delays It is recommended that the communication is started with no FEC OFF and onc
137. using the NAVI encoder and its button to confirm Some of these modes require advanced configuration such as the specification of IP ports etc or an auxiliary external STUN server that can be done in the NAT gt STUN submenu specifying its address host name IP and port following the same procedures explained to enter IP addresses in the Ethernet section 3 7 5 3 MAINTENANCE submenu MATH TENANCE EW VERSIONS DATE TIME TESTS BUZZER off DEFAULT SETTINGS e FW VERSIONS This menu can be accessed to check for informative purposes only the currently installed firmware versions with their dates of the different programmable devices within the unit CPU DSP and FPGA The audio processor code is included in the CPU firmware file If following indications from AEQ SAT any device version must be updated this operation must be done by using the equipment s Web Server as explained in section 7 1 of this manual NOTE Firmware updates dont modify unless explicitly stated by AEQ the current configuration of the unit and in particular its IP setup e DATE TIME It is possible to change the date and time of the unit by selecting this option In order to adjust the date select the first line and press the encoder button to change the field day month year and turn the encoder knob to increase or decrease the value When the year value is entered and the encoder button is pressed again no field will b
138. virtual circuit as the preceding ones Therefore the order of arrival of the packets to their destination is controlled and ensured e Datagram Each packet is routed independently from the rest Thus the network cannot control the path followed by the packets nor ensure the order in which they reach their destination B1 2 2 Advantages e In case of error in a packet only that packet will be resent without affecting other packets that arrived without errors e Interactive communication Limiting the maximum packet size ensures that no user can monopolize a transmission line for very long microseconds which means that packet switching networks can handle interactive traffic e Packet switching increases network flexibility and profitability e The pathway a communication takes can be altered from one moment to the next for example in case one or more of the routers breaks down I AEQ PHOENI X ALI O e Theoretically priorities can be assigned to the packets in a given communication Thus a node can select from its queue of packets waiting to be transmitted the ones that have higher priority B1 2 3 Disadvantages e Greater complexity of the intermediate switching devices which need to have higher speed and greater calculating capacity to determine the appropriate route for each packet e Packet duplication If a packet takes too long to reach its destination the receiving device may conclude that it has been lost in
139. which case it will send a packet retransmission request to the sender which gives rise to the arrival of duplicate packets e If the routing calculations account for an appreciable percentage of the transmission time the channel throughput useful information transmitted information decreases e Variations in the mean transit delay of a packet in the network Parameter known as jitter B2 IP protocol The Internet Protocol IP is a non connection oriented protocol used both by the origin and the destination in data transmission over a switched packet network The data in an IP based network are sent in blocks known as packets or datagrams in the IP protocol these terms are used interchangeably In particular in IP there is no need for configuration before a device attempts to send packets to another with which it has not communicated previously The Internet Protocol provides an unreliable datagram service called UDP User Datagram Protocol also known as best effort a phrase that expresses good intentions but offers few guarantees IP does not offer any mechanism to determine whether a packet reaches its destination and only provides security by means of checksums to cover its headers and not the transmitted data For example since it gives no guarantee that the packet will reach its destination it could arrive damaged in the wrong order with respect to other packets duplicated or simply not arrive If reliability is n
140. witching versus packet switching cccccceseeceeceeeeeeeeeeeeeeeaeeeesaeeeeeseeeeeesaeeeeeens 76 B1 T Circuit SWHCNINO DEE 76 ES Mgt acdc FAV ALAC EE 76 B1 1 2 Disadvantages EE T1 3 AEQ PHOENI X ALI O ENER Packet switching EEN 77 B1 2 1 Switching modes A TT B1 2 2 Advantages EE 77 B1 2 3 Disadvantages ceecccceeccceeeeceeeeceeeeeeeeeeseeeeseeeeeseeeessaeeesseeeesaneesaeeeesees 78 SER maze G EE 78 E ME ACS SO DE 19 B2 25 Unicast VS E e 19 Dorr ere EE 80 B3 1 Default PHOENIX ALIO Configuration cccccecececceeseeeeeeseeeeeeeseeeeeeeeeeeesaeeeeens 81 BU NOLO E 81 BA Te WV OR d leie Kr viel 81 B4 2 Possible work scenarios ccccscccsscecsscecscecaueeceeecueeceuceceusesageesaeeseeeeseeeseueessaeess 83 B4 3 PHOENIX ALIO default SIP Configuration ccccccccecseeeeeeeeeeeeeeaeeeeeeaeeeeeseeeeeeas 83 Ba TUN LOGON EE 84 APPENDIX C PORTS USED BY PHOENIX EQUIPMENT ccccseeeeeeeeeeeessseeeeeeeeeenneeeeees 86 APPENDIX D APPLICATION NOTES GUIDE 0 eee cestecseeeeenseeeeeneeeeeenseeseenseeseenseessoeneeees 87 D1 Internet connection using standard Cable ACCESS ccccceecccseeeeceeeeeaeeeeaeeeeeeeeseeeesees 87 Applicaton Noe FAN OP EE 87 Application aler LEE 87 Application note NA LEE 87 leiere inner LR RE 87 Application Nn te LEE 87 PROMI ALON TOUS ANGE E 87 D2 Special applications using different kinds of Internet physical a

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