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Remote Extensions .. various issues.


Carl Johnson

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We all know about NAT and its unfreindly nature concerning VOIP .. that said I am having a bit of an odd issue that just does not make sense as the SBC should resolve this, I think?

 

Polycom (not the greatest for external applications) .. will establish the SIP session and register as well as initiate a call but the audio (RTP) coming back from the PBX is always addressed to the wrong IP? Any thoughts on this does not seem to be an issue with a softphone and other hardphones in the same external network .. just the Polycom?

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We all know about NAT and its unfreindly nature concerning VOIP .. that said I am having a bit of an odd issue that just does not make sense as the SBC should resolve this, I think?

 

Polycom (not the greatest for external applications) .. will establish the SIP session and register as well as initiate a call but the audio (RTP) coming back from the PBX is always addressed to the wrong IP? Any thoughts on this does not seem to be an issue with a softphone and other hardphones in the same external network .. just the Polycom?

 

 

Just wonder what Polycom version are you running?

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We all know about NAT and its unfreindly nature concerning VOIP .. that said I am having a bit of an odd issue that just does not make sense as the SBC should resolve this, I think?

 

Polycom (not the greatest for external applications) .. will establish the SIP session and register as well as initiate a call but the audio (RTP) coming back from the PBX is always addressed to the wrong IP? Any thoughts on this does not seem to be an issue with a softphone and other hardphones in the same external network .. just the Polycom?

 

I don't think that makes a difference. The PBX generally sends the RTP to where it comes from, unless you disable that behavior (this is the "strict RFC mode" because the IETF had nothing better to do than say that RTP can actually come from different ports and even IP addresses). But Polycom and practically all SIP phone vendors that do have an interest to sell phones behind NAT send and receive on the same UDP port.

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The phone registers and shows its inside IP behind NAT, but the SIP packets are addressing it properly via the real-world address .. when sending RTP to the phone it will send it to the inside IP that is behind NAT which is obviously unroutable .. any ideas .. would a packet trace help?

 

For media-related issues, PCAP is the preferred way to see what is really going on. Maybe in this case the routing table of the PBX would also be useful.

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Hey Carl , long time !

 

 

I usually check the registration tab in the pbx and see what is has registered , either the 192.168.x.x or say 74.12.234.x ....

 

the SBC should take care of it unless the router is sending something it doesn't like .. then it's ugly stuff like STUN which may work but hate to use it long term ..

 

yori

 

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