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one way audio with SIP

Unanswered Question
Jan 27th, 2012
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For the past 3 years I have been using Linksys PAP2T phone adapter behind RV082 router with UDP port forwarding without any problems, until recently I started getting fast busy on all outgoing calls. Called the provider (ViaTalk) and was told they have to change the proxy setting on their end which fixed fast busy, but introduced ONE WAY audio on all incoming calls, however outgoing calls don’t seem to have this problem. ViaTalk’s solution to one way audio was changing SCTP port from 5061 to 5081 in the phone adapter, but as it made NO difference they blamed it on my router (which was working fine for the past 3 years) and hung up.


I was able capture phone adapter traffic using my laptop (nic in promiscuous mode), but to see the whole picture I think I need to do another capture between the cable modem and the router, just not sure what to capture.

The capture I got from behind the router looks like this. (66.147.235.3) sip:[email protected] communicates with my PAP2T (192.168.20.240) via 5061 and 5060 once connection is established, 192.168.20.240 communicates with 209.244.30.233 IP (G.711). Src Port: 16412 (16412), Dst Port: 60858 (60858) As I put my laptop with wireshark between the router and a cable modem, I am assuming 192.168.20.240 will not show up in the capture and the only IPs I can configure a filter for are 209.244.30.233 & 66.147.235.3?


Also I am guessing I will see something in the conversation with 209.244.30.233 in front on the router that I don’t see behind?


Thank you for your help

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Daniele Giordano Sat, 01/28/2012 - 00:55
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I think in a 2 possible issues:

- RTP ports blocked from your RV082

In this case you must add a new port forwarding to your PAP whit RTP ports used, tipically 16384-16482 but you can check under voice-sip menu.


- wrong contact address in SIP/SDP signalling

If the contact address in SIP/SDP is the private address 192.168.20.240, your provider can't established the RTP flow.

You can check the NAT Support Parameters under voice-sip menu and set the EXT IP using your pubblic ip address.

You can also set the NAT Settings under voice-Line menu.


Regards.

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