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Sometimes the SIP channel is connected to a server that is removed from the RTP sessions entirely. This would likely be the
case when two SIP devices are registered to the same (or sometimes even different) providers. The SIP channel would instruct
the devices to create RTP sessions between them, rather than to the provider. This is known as the “SIP Triangle”.

But more commonly, a SIP device is interested in making and receiving calls to and from the “old fashioned” public switch
telephone (PSTN) or “plain old telephone” (POTS) network, whether wired or cellular. In this case both the SIP channel and
the RTP sessions are made to a server at the Internet Telephone Provider, and the provider acts as a gateway for the voice call
to the “legacy network”. The user would be delivered a “real” phone number (DID for Direct Inward Dial) and the provider
would handle all the necessary VoIP <-> PSTN conversions. We’ll focus on this scenario from here on.

SIP Details

The technical details of SIP are widely available on the web for further research. But essentially, commands and formats are
provided to invite users to a call, accept calls, end them, and reject them. SIP also provides a mechanism to register and
authenticate with a server.

Another useful function in SIP is encoder negotiation. The SIP protocol can inform users of which encoders are supported on
each end of a session and in which priority. In this way, it’s easy to make decisions about which encoder to choose that will
be in common with both ends, and to reject calls if no common encoder is found.
Like RTP sessions, the SIP channel utilizes the UDP protocol by default. There is a specific port defined, 5060, as the default
“well-known” port over which SIP operates, although it can usually be configured to be different.

A single SIP channel can manage multiple RTP sessions simultaneously. In this way, only a single account needs to be
registered with the Internet Telephone Provider and a single SIP channel maintained, but multiple VoIP calls can be run
simultaneously. Whenever a call is initiated or dropped, a pair of RTP sessions is created or destroyed on the fly for each call.

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Содержание VH2

Страница 1: ...Product Manual ...

Страница 2: ...s 11 Contact Closures 12 Making Connections 12 III ConfIgurIng VH2 IP InformATIon 14 IV ConfIgurIng the VH2 Companion phone 16 Time setting 16 VH2 settings 16 V Telephone Connections 20 Introduction to SIP 20 Setting up a SIP provider or PBX 21 adVanCed sIp seTTIngs 22 under aCCounT InformaTIon 22 CodeC seTTIngs 22 under sIp seTTIngs 23 VH2 March 2017 ...

Страница 3: ...gaTeway seTTIngs 26 dIal prefIX 26 ouTgoIng enaBled 26 sIp Trunks 26 lIne assIgnmenTs 28 VI System Behavior 29 Auto Answer 29 Studio Audio I O 29 About Mix Minus 29 Caller Mix 29 AGC 30 Caller Ducking 30 Caller On Air Tone 30 Contact closures 30 Test Modes 31 VII Network Configuration Page 32 SYSTEM 32 IP Settings 32 Primary network 33 ...

Страница 4: ...g vh2 36 DIP Switch Settings 36 ON OFF Button ch 1 and 2 36 HOLD XFR ch 1 and 2 36 Indications 37 General Operation 37 Incoming calls 37 Outgoing calls 37 Transferring to Handset 37 Ending calls 38 Toolbox Call controls 38 XI About SIP 40 XII Information for IT Managers about VH2 53 InComIng serVICes 53 outgoIng serVICes 53 XIII CONTACT AND SUPPORT 54 ...

Страница 5: ...lic License Version 3 63 Lesser GNU Public License Version 2 1 68 GNU LESSER GENERAL PUBLIC LICENSE 68 Mozilla Public License Version 1 1 72 strace 77 XVI Programming your Polycom ip 331 for use with your PBX cloud based phone service or IP gateway 78 Digit map introduction 78 default digit map 78 Digit map time out 80 Editing your polycom digit map 80 Accessing the digit map 80 XVII SIP Trace 83 ...

Страница 6: ... selected when possible 3 VH2 utilizes a VoIP protocol called SIP which allows integration with existing and future VoIP gear e g sharing lines and using extensions from PBXs 4 VH2 can be easily interfaced to legacy PSTN ISDN T1 and E1 phone lines through the use of external gateway devices about VH2 audIo ProceSSIng VH2 s main function is to make your on air phone calls sound great From an audio ...

Страница 7: ...t from the VH2 This allows you to balance each caller on a separate console fader shows this arrangement Here two caller outputs are used but only a single mix minus feed is applied to the INPUT CH 1 port This mix minus must not contain any caller audio from either hybrid An example of how to create this type of mix minus is to use an auxiliary bus on your console deselecting both telephone input ...

Страница 8: ... output For simplicity we ll continue to use the legacy term for an on air telephone channel Some consoles have a dedicated send output for each fader associated with a caller Often the console will define these as telephone busses or channels In this case each hybrid gets its own send audio from the console each pre configured with an appropriate mix minus feed These mix minus feeds will be appli...

Страница 9: ...8 Finally VH2 is capable of behaving like two completely independent products for use in two unrelated studios This is shown below ...

Страница 10: ...ech to connect a feed to each return channel that is a mix of all important audio sources in studio mics automation carts other remote sources etc minus the caller Most modern consoles can do this easily If not other options exist and are discussed in the section Why Do I Hear Hear Myself Myself Section XIV in this manual Finally when configuring and connecting for using both send inputs the mix m...

Страница 11: ... configured to present all caller audio on this single output In digital AES3 mode both CH 1 OUTPUT and CH 2 OUTPUT are delivered here on left and right channel respectively AES3 available with 48KHz sampling rate only 4 CH 2 OUTPUT This analog output will deliver the audio from callers who are sent to channel 2 on air This output is disabled when in AES3 mode or when configured for a single calle...

Страница 12: ...ground The pinout is Tip Hold Audio in CH1 Ring Hold Audio in CH2 Sleeve Ground Do not feed this port with actively balanced outputs as it will ground one side of your active signal If using only a single Hold audio source connect the side of the feed to Tip and Ring and the Ground to Sleeve If using two different Hold sources connect CH 1 to Tip CH 2 to Ring and both grounds to Sleeve Even though...

Страница 13: ...ections At a minimum VH2 will need two audio connections and a network connection Levels of all analog audio I O is 0dBu 0 775V nominal This level will provide 20dB headroom before the clipping point Input audio is reflected on the front panel LED based peak meters as indicated in Clipping is indicated by the red LED on these meters VH2 Front Panel VH2 needs a network connection to be useful On VH...

Страница 14: ...VoIP phone in your system you will need to assign a Static IP address to the Ethernet port in order for the phone to find the VH2 About using the VH2 Companion Phone In a typical studio environment it is convenient to be able to use a telephone set in conjunction with a telephone hybrid This allows for outgoing calls to be placed calls to be answered and screened and for conversations to take plac...

Страница 15: ...results should look like this There are two ways to set the static IP information using Device Manager The rescue mode allows you to change parameters during the first five minutes of operation and is initiated using the Network Settings button on the right pane This mode is most useful if you ve forgotten the unit password The normal mode of changing any settings in VH2 is by clicking the Web Con...

Страница 16: ...you ll need to rescan and log in again or open a browser to the new IP address Alternately to use the Rescue Mode just click the Network Settings box on the right pane of Device Manager If the 5 minute timer has not run out a pop up menu will appear allowing you to reconfigure the IP address netmask Gateway and DNS information and save it away You will need to restart the VH2 for these changes to ...

Страница 17: ...ey 2 Status 2 Network 1 TCP IP Parameters Open a browser on your local computer and input the IP address of the phone into the URL bar You will be prompted to log in to the phone s web interface Default username is Polycom and default password is 456 Time setting To stop the annoying time display from flashing an incorrect time and date set a default SNTP server and time zone on the phone Navigate...

Страница 18: ...hentication User ID 1100 Authentication Password default 456 can be changed in Toolbox Label Line 1 Type Private Third Party Name none Number of Line Keys 1 Calls per Line 1 Under Server 1 Address VH2_ip_address remove Port 5170 Transport DNSnaptr Expires 100 Register 1 Retry Timeout 0 Retry Maximum Count 3 Line Seize Timeout 30 leave the rest of the fields at default Your Line 1 Page should read ...

Страница 19: ...er Identification Display Name Line 2 Address 1200 VH2_ip_address remove and Authentication User ID 1200 Authentication Password default 456 can be changed in Toolbox Label Line 2 Type Private Third Party Name none Number of Line Keys 1 Calls per Line 1 Under Server 1 Address VH2_ip_address remove Port 5170 ...

Страница 20: ...Timeout 0 Retry Maximum Count 3 Line Seize Timeout 30 leave the rest of the fields at default Your Line 2 Page should read like this but with your VH2 IP address Click the Submit button on the bottom of the page You are finished with the telephone setup ...

Страница 21: ... process can t be described here but the result of this process is that you will have access to certain credentials for that account These typically consist of Server Domain The IP address or URL of the server to which you are being registered Username The name that VH2 will use when logging into the service Password The password associated with your account for security purposes In addition sever...

Страница 22: ...each you VH2 does not need to know this number translation to the proper SIP channel happens behind the scenes at the SIP provider although often the DID and SIP account name are the same In Toolbox navigate to Channel Configuration VoIP Providers Add Provider SIP Provider The basic settings for a SIP provider are shown in the figure above First you should give your provider a unique name in the N...

Страница 23: ...nd is the most common codec used in VoIP When your SIP provider or PBX bridges incoming or outgoing calls to the legacy phone network it will use this codec So if you are using this mode primarily e g not taking or making any wideband calls on this provider this is often the best codec to use so no transcoding needs to occur The downside is that by modern standards G 711 uses a lot of network data...

Страница 24: ... the SIP provider will consider the registration connection lost and once lost how often to attempt to re establish The default values are usually best unless strictly required to be changed by your provider Register If you would like to save SIP provider entries for occasional use you can set them to be disabled here by setting this value to No No registration will be attempted until this setting...

Страница 25: ... Gateways allow you to use VH2 with traditional analog phone lines as well as T1 BRI and PRI ISDN and other legacy telephone trunks Gateways will convert these telephone channels to SIP style virtual phone lines You will need to find gateways that deliver FXO style channels on their telco side the ports on the gateway are designed to point toward the telephone service and not interface with teleph...

Страница 26: ...y VH2 but can be changed to any value you wish general seTTIngs Name Give your gateway a unique name aCCounT InformaTIon Username and Password Locally generated values that the gateway will use to register to VH2 sIp seTTIngs Address The IP address of the gateway Gateway Binding Port Automatically populated with an unused port Must be mirrored into the Gateway Settings ...

Страница 27: ...as guides for this process You may also come across certain PBXs that must deliver their extensions in this way In order for STAC VIP to receive these extensions a SIP trunk must be created instead of a provider account SIP Trunks also differ from normal SIP providers in several ways Rather than having STAC VIP pull the SIP channel from a provider a SIP trunking provider will push the channel to a...

Страница 28: ...the Destination Number field 2 Trunk Incoming Match This is the field where you enter the text that will be matched by the system to the incoming call If the Network Address option is chosen It s OK to leave the Trunk Incoming Match field blank this is the default configuration and the system will use the Server Realm field entry for the match Alternately you can input a different IP address to be...

Страница 29: ...vider s before you can use it You can have up to two providers and each provider can be assigned to one or both channels Channel assignment is done in Toolbox via Channel Configuration Channel Assignments The figure shows each channel assigned to a different provider Each channel could also be assigned to the same provider and the usual behavior is for calls to hunt between the channels first call...

Страница 30: ...CH 1 OUTPUT Caller 2 hears CH 2 INPUT and appears on CH 2 OUTPUT Note that none of these settings has any effect on whether callers hear each other That function is chosen in the next option About Mix Minus When we refer to send audio to the caller we re talking about the feed that is attached to the VH2 CH 1 INPUT and CH 2 INPUT This is the audio that the caller hears when on air It is essential ...

Страница 31: ... any line 3 Call On Air 1 2 A call has been placed on air and is active on either the CH 1 OUTPUT or CH 2 OUTPUT specifically 4 Call Ringing An incoming call is ringing on any configured line Inputs Momentary Latching The four contact closure inputs parallel the four front panel On Off and Hold Xfr buttons Since these buttons act as momentary toggles this is the default action of the contact closu...

Страница 32: ...be enabled e g CH 1 INPUT to CH 1 OUTPUT These are used in unit production tests and can also be used to troubleshoot general hookup issues Modes are also offered that generate a tone from the caller out ports Contact Closure Test Likewise enabling this option puts the contact closure feature in loopback mode with inputs directly driving outputs e g CH 1 INPUT to CH 1 OUTPUT LED Test Mode tests th...

Страница 33: ...2 via the SSH protocol to troubleshoot issues This requires a private keypair that we don t provide If you have security concerns about SSH you can disable it by setting this option to No System Clock VH2 maintains a network connection to an NTP server that delivers the time of day information for use by system logs The specifics of that function can be changed here The default settings allow for ...

Страница 34: ...further configuration of VH2 is possible To change any setting you must first apply a factory reset to the VH2 which will wipe all its settings including all your VoIP account info and static IPs Turning services off also disables your VH2 s ability to sync with the Comrex Device Manager until a factory reset is performed Port 80 This is the port that runs web and XML services ...

Страница 35: ...he web or the keys on the phone If you wish to change this password from the default you can let VH2 know the new value in this field Comrex recommends changing this password Changing the Password on the admin account It is highly recommended to change the password on VH2 to prevent undesired configuration changes This is done under Security Accounts Admin in Toolbox Note that once the password is...

Страница 36: ...er software to login to VH2 Select Device Reset to Factory Defaults to issue the reset command 2 If the password is not known you must issue a hardware factory reset This is done via the following sequence 1 Put dip switch 7 8 up 2 Press the reset button once wait until the CH 1 CH 2 and Power LED flash red and green 3 Put dip switch 7 8 down 4 Press the reset button once ...

Страница 37: ... on air and caller audio is routed via the selected behavior settings Pressing this button when a call is active drops the call whether on air or on hold HOLD XFR ch 1 and 2 This is a toggle that places an active call on hold from the on air state The caller will be removed from the main audio ports and hear only the audio presented to the VH2 on hold input Pressing this button while an incoming c...

Страница 38: ...t connectivity detected Green System has IP address on network General Operation Incoming calls Calls will ring the handset if used and blink the ON OFF indicator for the channel with the incoming call Calls can be answered via the handset or placed directly on air or on hold via the front panel button Calls on the handset can be put on air or on hold via the same buttons Outgoing calls Place call...

Страница 39: ...press the ON OFF button to end the call From the handset simply hang up Toolbox Call controls When logged into the web based Toolbox utility the last item listed is Control The status of both Channel one and Channel two are presented here as well ...

Страница 40: ...either by placing the calls on air on hold transferring to the handset or dropping the call This allows you to control the VH2 channels without having to physically push the buttons Additionally it will show the status at the top of the screen for both channels ...

Страница 41: ...lternately the VoIP lines can be delivered from an upstream PBX The end user gear is a specialized VoIP telephone or software running on a PC or mobile device that performs the same functions The Comrex STAC VIP is a sample of a device designed to interface with VoIP service It can handle six or twelve calls simultaneously and provide the typical screening audio processing and control functions ex...

Страница 42: ...he web This is because TCP has quite a bit of overhead in terms of data and can easily add time delays if packets get corrupted VoIP and other real time communication protocols use UDP which is a much simpler delivery method There is no error correction or resending available at the native UDP layer UDP is sometimes referred to as the send and pray method since the network provides no guarantees o...

Страница 43: ...ncept of how a gateway router provides translation services to the Internet is extremely important in the field of VoIP if only because it causes so many headaches Known as Network Address Translation NAT it s easiest to use a diagram to illustrate a typical gateway scenario describing a user on a LAN accessing a web page at comrex com For this illustration we ll ignore the concepts of DNS and URL...

Страница 44: ...sending it along to the LAN In reality NAT is more complex than this changing port numbers as well but we ve kept the concept to the bare basics to outline why NAT hurts VoIP NAT provides for many benefits including address reuse and basic security This security exists because packets that arrive from the public Internet without being requested from within the LAN will be discarded But it s this s...

Страница 45: ... is a protocol layer that exists within a UDP packet specifically designed to transfer audio and video media with low delay RTP consists of a header that is applied directly after the UDP header in the packet followed by a media payload which consists of the actual encoded audio of a VoIP call ...

Страница 46: ...his is the choice of encoder within the system used to compress digital audio so it uses less network capacity Encoders like MP3 and AAC are common in that world You ll see the VoIP industry use the term codecs for this function But because broadcast transmission devices are also termed codecs we ll reserve it to describe hardware and use encoders to describe compression algorithms VoIP has its ow...

Страница 47: ...s and VoIP devices support G 722 Opus Efforts are increasing at combining the worlds of VoIP and web services Many web audio services have standardized on Opus an encoder that delivers near CD quality audio with low delay As these efforts continue users can expect to find more support for the Opus codec in VoIP devices and networks All Comrex codecs and the STAC VIP phone system support Opus Other...

Страница 48: ... example would be a purely IP PBX In this case the PBX maintains a SIP channel to an Internet Telephone provider on its WAN port It also maintains several SIP connections over its LAN to telephone extensions Because the protocol used in these links is identical it provides for a lot of flexibility For example if need be the telephone extensions could register directly with the provider bypassing t...

Страница 49: ...etails of SIP are widely available on the web for further research But essentially commands and formats are provided to invite users to a call accept calls end them and reject them SIP also provides a mechanism to register and authenticate with a server Another useful function in SIP is encoder negotiation The SIP protocol can inform users of which encoders are supported on each end of a session a...

Страница 50: ... to receive information about new incoming calls from the provider the user end must keep the SIP connection or binding open through the NAT router to prevent it from terminating the binding and blocking incoming traffic It does this by sending periodic updates even when no changes are being made to any calls The interval of these updates is usually adjustable but must be shorter than the timeout ...

Страница 51: ...h its security layer It s not aware this session has been requested so it s blocked by default This usually results in a one way connection where no audio can be heard on the SIP user end of the call ALG to the rescue This scenario has become common enough that router and firewall manufacturers have started to address it The solution is called SIP ALG for application layer gateway and has been bui...

Страница 52: ...tion stays open on UDP 5060 between the user and the service provider 2 Separate and multiple RTP sessions are established in each direction for calls 3 Routers and firewalls interfere with these RTP sessions by design but ALGs built into these devices can help PBXs So far we ve discussed SIP connections to outside or cloud VoIP providers But many times the user already has a SIP PBX on premises w...

Страница 53: ... will create the RTP channels in the same way as for incoming calls Hunting Of particular interest to broadcasters who take lots of calls simultaneously is hunting behavior or the way the system behaves toward simultaneous incoming calls Keep in mind when an incoming call is in the ringing state there are only status messages exchanged over the SIP connection no actual audio is being transferred T...

Страница 54: ...ion the update data itself must have a valid cryptographic signature from Comrex or else it is rejected In order for the unit to be remotely updated TCP port 8080 must be forwarded to the device Alternately updates can be initiated from any local PC using the Device Manager The device can support connection to a SIP trunking service which would require incoming service on a single UDP SIP port usu...

Страница 55: ...Devens MA 01434 USA Technical Support is available Monday Friday 8 30AM 5PM EST 1 800 237 1776 North America 1 978 784 1776 International 1 978 784 1717 FAX email techies comrex com Product manuals and firmware updates available on the web at http www comrex com ...

Страница 56: ...speaker s equipment rather than on the far end of the call This is because humans have a very hard time handling even the smallest delay in this sidetone signal In testing we find that any delay over around 10mS starts to have an effect called slapback where the speaker is unable to maintain conversation and begins to halt and stutter Even in old fashioned analog telephone circuits it s possible t...

Страница 57: ...solution here is mix minus a term used for a special mix of audio that explicitly excludes one source the audio coming from the place the mix minus is being sent To put it another way mix minus is the entire studio mix minus one audio source So how do we create this special audio mix On modern studio systems this is usually well defined and easy to do Many consoles feature channels dedicated to te...

Страница 58: ...llow telephone callers to appear on one of two outputs and therefore on two separate console faders In this circumstance you often have a choice of delivering a single mix minus with neither of the telephone audio sources present or two distinct mix minus feeds In the case of two feeds it s important to note that mix minus A must include the caller audio B and vice versa The following figure illus...

Страница 59: ... no resources to do so investigate the Comrex Mix Minus Bridge product It s an easy way to deliver up to six distinct mix minus feeds e g for two telephone channels two audio codecs a two way radio system and an RPU return channel simultaneously while sacrificing only one console auxiliary bus ...

Страница 60: ...HE SERVICES PROVIDED BY OR ON BEHALF OF COMREX OR II FOR ANY MATTER BEYOND COMREXS OR ITS LICENSORS REASONABLE CONTROL COMREXS AND ITS LICENSORS MAXIMUM LIABILITY FOR DAMAGES OF ANY KIND WHATSOEVER ARISING OUT OF THIS AGREEMENT SHALL BE LIMITED TO THE PURCHASE PRICE PAID BY YOU FOR THE SOFTWARE EXCEPT WHERE NOT PERMITTED BY APPLICABLE LAW IN WHICH CASE COMREXS LIABILITY SHALL BE LIMITED TO THE MIN...

Страница 61: ...are Also for each author s protection and ours we want to make certain that everyone understands that there is no warranty for this free software If the software is modified by someone else and passed on we want its recipients to know that what they have is not the original so that any problems introduced by others will not reflect on the original authors reputations Finally any free program is th...

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Страница 77: ...ode for Your Modifications Dropbear Dropbear contains a number of components from different sources hence there are a few licenses and authors involved All licenses are fairly non restrictive The majority of code is written by Matt Johnston under the license below Portions of the client mode work are c 2004 Mihnea Stoenescu under the same license Copyright c 2002 2008 Matt Johnston Portions copyri...

Страница 78: ...Paul Kranenburg Copyright c 1993 Branko Lankester Copyright c 1993 Ulrich Pegelow Copyright c 1995 1996 Michael Elizabeth Chastain Copyright c 1993 1994 1995 1996 Rick Sladkey Copyright C 1998 2001 Wichert Akkerman All rights reserved Redistribution and use in source and binary forms with or without modification are permitted provided that the following conditions are met 1 Redistributions of sour...

Страница 79: ...numbers can make up extensions within your company How will your IP phone know that you have completed entering the numbers and initiate the call if it doesn t know how many numbers are in your extensions The answer digit maps When dialing a number using the Polycom IP 331 with the handset still in the cradle i e when using the speaker feature or a headset you have the ability to indicate you have...

Страница 80: ...n has the digits 1 and 1 This matches the sequences of 211 311 911 and will then dial those digits immediately once a match has been found 0T This will match the digit 0 and after a time out of 3 seconds defined in the Digit map Time out will send the 0 digit to the PBX The T is what specifies that a time out will be used to determine when to send the digits to the PBX 011xxx T This will match any...

Страница 81: ...u must set the digit map time out to 3 3 3 3 7 because the entry for which we are interested in extending the time out is the fifth entry in the digit map as defined by the vertical bars If you have too many digit map time out values they are ignored If you have too few values they default to 3 seconds Editing your polycom digit map Although it is recommended to have your System Administrator edit...

Страница 82: ...map entry The following code shows the digit map needed if your PBX requires the digit 9 to be entered before allowing any outbound calls 9 2 9 11 0T 9011xxx T 9 0 1 2 9 xxxxxxxxx 9 2 9 xxxxxxxxx 2 9 xxxT The digit 9 has been added before the two strings responsible for local and long distance calling in North America International calls as well as the 2 9 11 numbers for many standard 3 digit outb...

Страница 83: ...xxxxxxxxx 9 2 9 xxxxxxxxx 2 9 xxxT editing the digit map for long distance calling from outside the us The following is an example for placing international calls from outside the US The US uses the prefix 011 for placing international calls but many countries have different prefixes For countries that require 00 for international calling overwrite edit the digit map code with the following 2 9 11...

Страница 84: ... and from the VH2 or select one of your SIP Providers from the drop down menu to perform a packet capture for one specific provider Selecting All will provide unfiltered data including broadcast data on your network There can be a lot of information in this file that may not be necessary Selecting the individual providers will give you a more precise set of information Press Save Setting once sele...

Страница 85: ...essed a new button titled Download Trace becomes available Press this to download the packet capture file a pcap extension This will start an http download to your browser To review this file we recommend using Wireshark a free and open source packet analyser available online ...

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