Using OBi Voice Gateways with SIP Providers
RonR:
Quote from: murzik on March 30, 2011, 06:22:27 am
What I meant if sipborker and voxalot are work for you, because what I get is just one way audio.
Especially calling any peers through sipbroker.
I agree with oleg that it's a NAT/RTP issue. I haven't had any problems so far, but I also forward all applicable SIP and RTP ports to the appropriate devices as well as use a STUN server whenever possible. VoIP audio is a problem waiting to happen when NAT is involved with SIP/RTP.
A good test to see if your router is possibly the culprit is to simply bypass it, if possible. Connect the OBi directly to your Cable/DSL modem and see if your audio issues go away. If they do, your router is definitely a suspect. Make sure you reboot both devices after connecting them directly to each other.
JohnLennon:
Great solution!
oleg:
Quote from: RonR on March 30, 2011, 09:58:14 am
A good test to see if your router is possibly the culprit is to simply bypass it, if possible. Connect the OBi directly to your Cable/DSL modem and see if your audio issues go away. If they do, your router is definitely a suspect.
Bypassing router you eliminate NAT and thus the need in STUN detection. Audio issues may go away. But how it makes "router a suspect" ??? That only proves that router has NAT inside and that router separates local network from the internet. Anybody doubts? ;)
Normally it is responsibility of SIP device to detect internet address / port and send them to another SIP device. STUN protocol serves that purpose. Routers usually do not care about SIP protocol.
There are a few rare exceptions though:
- some NAT implementations may be practically impossible to traverse even using STUN.
- some rare routers (VOIP routers, SIP proxies) may recognize SIP protocol and substitute local address / port with internet address / port.
RonR:
Quote from: oleg on March 30, 2011, 07:25:09 pm
Audio issues may go away. But how it makes "router a suspect" ???
The very first paragraph of the article linked below does a much better job of describing the problem than I ever could:
"The very first thing to note is that SIP was NOT designed to work with NAT. There are subsequent standards, hacks, workaround, kludges etc. to try and make it work but the original SIP designers somehow deemed it beneath them or put it in the too hard basket to bother coming up with a proper solution (there is not one instance of the string “NAT” in the whole SIP RFC)."
http://sipsorcery.wordpress.com/2009/08/05/nat-rtp-and-audio-problems/
If taking your router of the loop eliminates your audio issues, there's an excellent chance your router has a NAT implementation that's not SIP/RTP friendly. I've got a number of older routers in the store room that fit that description. Newer routers tend to do a lot better job, but they're still not all created equal and depending on the service provider's SIP implementation, one router may work in a particular situation but not in another. IOW, it's a crap shoot.
oleg:
Indeed, SIP was not designed to work with NAT. But most contemporary SIP implementations (including all Linksys ATAs) use STUN and successfully traverse most NATs.
Try another experiment - replace OBi110 build 2101 with build 1892, make sure STUN enabled and configured. There is an excellent chance that audio issues go away with the same router (unless router belongs to first of two exceptions I mentioned above). According to your logic that proves OBi110 build 2101 "is definitely a suspect". Here I agree with you :)
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