Comrex BRIC-Link II, Product Manual

Page: 90 / 104

SIP registra on can be very simple with some servers, and others can require more advanced se ngs, which are
described in the
Advanced Topics sec on below.
MAKING REGISTERED SIP CALLS
When registered, calls made using an EBU3326/SIP pro fi le behave di ff erently than normal. The address fi eld,
regardless of whether it is a SIP URI or an IP address, is forwarded to the server. No connec on a empt is made
un l the server responds.
If the server accepts the address, the call will be a empted. If not, an error message will appear in the status line.
Some examples of reasons for call rejec on by a server are:
• The server does not support direct connec on 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 applica ons. There are
inevitably situa ons where the defaults won’t work, however. Comrex has provided some advanced op ons that
can help. These op ons are located in the
Systems Se ngs and can be made visible by enabling Advanced Op ons :
• IP Port - Universally, SIP connec ons are supposed to use UDP port 5060 to nego ate calls between
devices (and between servers and devices). Note that this is only the nego a on channel—actual audio
data is passed on the RTP ports. Changing this port number will change which incoming ports are used
to ini ate connec ons and to which ports connec on requests are sent. The change must be made on
both devices, and this change will essen ally make your codec incompa ble 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 nego ated at the beginning of a call (over the IP port),
this port may be changed without breaking compa bility. Note that many SIP standard devices use port
5004 for this func on. Due to the nego a on, it is not important that these numbers match on each
end. Changing this port to 5004 can actually have an adverse e ff ect, as 5004 is the default port for other
services on Comrex codecs.
• Public IP Override - See the SIP Troubleshoo ng for more informa on on this op on.
• Use STUN Server - See SIP Troubleshoo ng for more informa on on this op on.
• SIP Proxy Keepalive - Only applies to Registered mode. This variable determines how o en the codec
“phones home” if registered with a SIP server. It’s important that the codec periodically “ping” the server,
so the server can
fi nd the codec for incoming calls. It can be adjusted primarily to compensate for fi rewall
routers that have shorter or longer binding mings, i.e., the router may have a tendency to “forget” that
the codec is ready to accept incoming calls and block them.