Making Registered Sip Calls; Advanced Ebu3326/Sip Topics - Comrex ACCESS MultiRack Product Manual

mAkIng regIstered sIp cAlls

When registered, calls made using an EBU3326/SIP profile behave differently than normal. The address field,
regardless of whether it is a SIP URI or an IP address, is forwarded to the server. No connection attempt is made
until the server responds.
If the server accepts the address, the call will be attempted. If not, an error message will appear in the status line.
Possible reasons for call rejection by a server are numerous. Some examples are:
• The server does not support direct connection to IP addresses (if the address is in this format).
• The server does not recognize the address.
• The server does not forward calls beyond its own domain.
• The server does not support the chosen codec.
• The called device does not support the chosen codec.
• The address is a POTS telephone number, and POTS interworking is not supported.
• The address is a POTS telephone number, and no credit is available (most services charge for this).

AdVAnced ebu3326/sIp topIcs

The basic entries provided will allow support for the vast majority of EBU3326/SIP-based applications. There are
inevitably situations where the defaults won't work, however. Comrex has provided some advanced options that
can help. These options are located in the Systems Settings and can be made visible by selecting the Advanced box:
• IP Port - Each MultiRack instance uses a different UDP port for SIP negotiation. Instance #1 uses the
well-known 5060 port. Instance #2 uses 5062, #3 uses 5063 etc. For non-registered incoming calls, it's
important to know this port number, because it needs to be added to the calling unit's outbound IP
address (e.g. 192.168.25.25:5064).
• RTP Port - This is one of two port numbers used for audio data transfer; the port number directly above
this is used as well. Because this port number is negotiated at the beginning of a call (over the IP port),
this port may be changed without breaking compatibility. Instance #1 on MultiRack uses UDP port 6014
and 6015 for SIP RTP. Instance #2 uses a port 10 steps higher of 6024 and 6025. Instance #3 uses 6034 and
6035, etc. Often, for incoming calls to work correctly behind routers and firewalls, these ports must be
forwarded to MultiRack. See the section on SIP troubleshooting for more info.
• Public IP Override - See the SIP Troubleshooting section for more information on this option.
• Use STUN Server - See the SIP Troubleshooting section for more information on this option.
• SIP Proxy Keepalive - Only applies to Registered mode. This variable determines how often the codec
"phones home" if registered with a SIP server. It's important that the codec periodically "ping" the server,
so the server can find the codec for incoming calls. It can be adjusted primarily to compensate for firewall
routers that have shorter or longer binding timings, i.e., the router may have a tendency to "forget" that
the codec is ready to accept incoming calls and block them.
• SIP Domain - This only applies to Registered mode. It's the name of the network controlled by the SIP
server. This parameter must be passed by the codec to the server. Under most circumstances, this is the
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