Hi, I am troubleshooting call quality issues for a site using g729. I followed https://wiki.wireshark.org/HowToDecodeG729 and https://wiki.freepbx.org/display/PC/How+to+decode+and+playback+G.729+audio+streams (Followed this one worked for my side of the stream) For my side of the RTP stream I was able to get everything decoded and it came through fine. For their side of the RTP stream it came out to pure static. This isn't what I heard during the actual call. I started the pcap before the call so wireshark gets the SIP signalling. Wireshark sees both rtp streams clearly. Any insight shared is appreciated. edit: The audio file I got from the other rtp stream ended up about twice as long as my side's rtp. asked 12 Oct '17, 13:42  hIGH_aND_mIGHTY edited 12 Oct '17, 14:34 |
One Answer:
I would say this recent question is relevant. To me it seems that the decoder recommended above has some issues, causing it to interpret some streams worse than others. The decoder which is part of Wireshark's RTP player provides better results, but to date it is only available in a development version of Wireshark (2.5.x). answered 13 Oct '17, 12:37  sindy |
This took care of my question. Thank you for responding.
@hIGH_aND_mIGHTY
I've converted the comment from @sindy to an answer. As it seems to have solved your issue, please accept the answer for the benefit of other users by clicking the checkmark icon next to the answer.