SIP as RTMP issues

[Yury Pogrebnyak]

I’m trying to forward data from a SIP call to my RTMP server. Everything seems to work except I don’t see the stream on my server side. WCS is successfully connecting to a call, and also judging from the logs (please see attachment) it is able to publish RTMP stream. My codecs in flashphoner.properties are set to ulaw.

However, I don’t receive the stream. I’m using nginx rtmp module, and both my access.log and error.log are clear. I don’t have any security settings on rtmp server and also able to receive streams from different sources. What could possibly be the issue?

Thanks in advance!

 

[Max]

Please provide following logs and files:

/usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner.log
/usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner_manager.log
/usr/local/FlashphonerWebCallServer/conf

and pcap file of traffic dump

tcpdump -s 4096 -w log.pcap

You can send the archive to logs@flashphoner.com
If your SIP endpoint has auto-answer, please provide SIP details, we will check this call from our server.

 

[Yury Pogrebnyak]

Here's my archive. Also duplicated to logs@flashphoner.com

 

[Max]

There is a configuration issue: codecs not to be used in SIP as RTMP case should be listed in 'codecs_exclude_sip_rtmp' not removed from 'codecs'.

For example,
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =opus,alaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv