Comrex BRIC-Link Product Manual page 80

SIP registration can be very simple with some servers, and others can require more advanced settings, which are
described in the Advanced Topics section on the following page.
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.
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 - Universally, SIP connections are supposed to use UDP port 5060 to negotiate calls between
devices (and between servers and devices). Note this is only the negotiation channel; actual audio data is
passed on the RTP ports. Changing this port number will change which incoming ports are used to initiate
connections and to which ports connection requests are sent. Obviously, the change must be made on
both devices, and this change will essentially make your codec incompatible with industry-standard VoIP
devices.
• 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. Note that many SIP standard devices use port
5004 for this function. Due to the negotiation, it is not important that these numbers match on each
end. Changing this port to 5004 can actually have an adverse effect, as 5004 is the default port for other
services on Comrex codecs.
• 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
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