91
Use STUN Server
- See the next section,
SIP Troubleshooting
for more information.
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
- [only applies to
Registered
mode]. This is 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 same as the server/
proxy address, and if this field is not populated, that is the default. If, for some reason, the domain is different than
the server/proxy address, then this field is used.
SIP troUBlEShootIng
In a nutshell, SIP establishes a communication channel from the calling device to the called device (or server)
on port 5060. All handshaking takes place over this channel, and a separate pair of channels is opened between
the devices: one to handle the audio and the other to handle call control. The original communication channel is
terminated once the handshaking is complete. Note that firewalls must have all three ports open to allow calls to
be established correctly. Also, port forwarding may be required to accept calls if your codec is behind a router.
The main area where SIP complicates matters is in how an audio channel gets established once the handshake
channel is defined. In the common sense world, the call would be initiated to the destination IP address, then the
called codec would extract the source IP address from the incoming data and return a channel to that address. In
fact, that’s how the default mode of Comrex codecs work, and it works well.
But SIP includes a separate “forward address” or “return address” field, and requires that a codec negotiating a call
send to that address only. This is important in the case of having an intermediate server. And this works fine as long
as each codec knows what its public IP address is.
oUtgoIng Call ISSUES
A unit making an outgoing call must populate the ”return address” field. But any codec sitting behind a router has
a private IP address, and has no idea what the public address is. So, naturally, it will put its private IP address (e.g.
192.168.x.x
style) address into that “return address” field. The called codec will dutifully attempt to connect to that
address and undoubtedly fail, since that can’t be reached from the Internet at large.
InCoMIng Call ISSUES
Incoming calls to codecs behind routers are complicated by the fact that ports on the router must be forwarded to
the codec. In the case of SIP, this must be three discrete ports (For Comrex codecs these are UDP 5060, 5014 and
5015)<6014 and 6015 with 3.0 firmware>. And since even the “forward address” is negotiated in SIP, the incoming
unit is likely to populate the “forward address” field with its private address as well.
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