Hi all,
Our current setup is running 18.16.0 and pjsip with alaw+ulaw codecs.
Not too long ago we added to the endpoints an rtp_keepalive
of 4
which helped resolve some issues of silence and it worked very well to add CN.
However we are getting issues still where calls do drop when silent and sending out to the PSTN. We contacted this particular provider who said the reason is whilst CN packets are generated they are effectively ignored because we have not negotiated CN.
Checking the sdp we can see that whilst others add the following line it is not added from the PBX a=rtpmap:13 CN/8000
which is comfort noise for ulaw+alaw at 8000hz.
From the rfc spec 5.1 it does appear to be mandatory for use of CN with 8000hz:
Having looked through various posts and the documentation I cannot see anything related to getting this line added in the sdp. Other options we are thinking about include, though none are preferable:
- Use our proxy to manually add this line
- Pick a different codec
- Upgrade (but can’t see any notes about this issue)
- Find a way to add the line in Asterisk dialplan or similar (however not editing the codebase)
From what I gather this is more pjsip than Asterisk. Is there anything we are missing here, or is it something that pjsip offers but has not been exposed to Asterisk?
Many thanks!