Asterisk 18 server sends RTP streams with same SSRC, but wrong sequence numbers

I am trying to get some help on what my next steps should be to solve the following issue. Agents will call Asterisk and connect as a call to AgentLogin(). While connected, an AMI command will dial and enter into an interface the outbound call that they want to make. Dialplan will place the call and connect to the agent.

After about 3 rings the call goes silent (my guess is that some picked up or the VOIP phone dropped RTP packets). Audio recording can hear both parties. Agent can’t hear called person. Called person can hear agent. This does not happen all time, just occasionally. Hard to reproduce. The vendor that we use for our Softphone SDK, says that:

“Asterisk 18 server sends RTP streams with same SSRC, but wrong sequence numbers”

Our issues started when we moved to Asterisk 18 in May 2021, but it picking up steam now. I am not 100% that its an Asterisk issue.

We have gotten this issue on: 18.3, 18.4, 18.5.1

Does any one have any thoughts on my next steps? Or how I can fix the issue? …

Thanks
Jeff

Asterisk’s handling of SSRC has always been a bit weird, although, as I recall it, it is the timestamp that can suddenly jump. It expects systems to treat a marked frame as, effectively being a source change, rather than a good, but optional, place, to reset jitter buffers.

Most systems cope well with this, and the only issue I’ve personally encountered was with stuttering audio, rather than silence.

It is quite a long time since I last analysed this, so the issue I’m describing here may have been addressed. It’s not something that would have appeared in Asterisk 18.

I would suggest capturing the RTP with tcpdump and analysing with wireshark. That should show up discontinuities in sequence numbers and timestamps, and SSRC changes.

You should also specify which version of 18.

We have encountered this issue on all these versions:
18.3, 18.4, and 18.5.1

Thanks
Jeff

Then you’d need to look at a packet capture as David said.

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