Call via SIP disconnecting after around 1 minute

[Luca]

Hi

I am trying to implement SIP calls to a TWILIO SIP domain server that redirects the call inside a conference call.

The call disconnects after around 60 seconds and cannot understand why.

Please let me know if you can help

Thank you
Luca

 

[Max]

Good day.
Please reproduce the problem and collect a report as described here including client debug logs and traffic dump, then send it using this form. If the archive size exceeds 30 M, place it to Google Drive, OneDrive etc and send the link using Comment field of the form.

 

[Luca]

Good day.
Please reproduce the problem and collect a report as described here including client debug logs and traffic dump, then send it using this form. If the archive size exceeds 30 M, place it to Google Drive, OneDrive etc and send the link using Comment field of the form.

I have submitted the pcap log file via the form provided, please let me know if anything else is needed

Thank you
Luca

 

[Max]

Unfortunately the traffic dump seems to be corrupted:

Also, you've not provided a full report including server logs and client debug logs as mentioned in this post. So you should do the following:
1. Enable client debug logs

client_log_level=debug

2. Restart WCS
3. Start traffic dump collection on server

tcpdump -npi any -B 10240 -w log.pcap

4. Establish a call, reproduce the problem
5. Stop traffic dump collection
6. Collect the report using report.sh script: Getting logs with report.sh script
7. Pack traffic dump with report archive.
Then, send the resulting archive using this form.

 

[Luca]

Unfortunately the traffic dump seems to be corrupted:
View attachment 3353
Also, you've not provided a full report including server logs and client debug logs as mentioned in this post. So you should do the following:
1. Enable client debug logs

client_log_level=debug

2. Restart WCS
3. Start traffic dump collection on server

tcpdump -npi any -B 10240 -w log.pcap

4. Establish a call, reproduce the problem
5. Stop traffic dump collection
6. Collect the report using report.sh script: Getting logs with report.sh script
7. Pack traffic dump with report archive.
Then, send the resulting archive using this form.

Thank you for the response, I have resent the report that was produced using the report.sh command

Please let me know if anything else is needed for the investigation

Thank you
Luca

 

[Max]

We checked the report.
Both logs and traffic dump show there is no media traffic from browser. The call is finished by RTP activity:

That's because you've starting a call using REST API /call/startup. In this case, a special media traffic generator must be enabled:

generate_av_for_ua=all

Also, for audio only calls we recommend to exclude all the video codecs:

codecs_exclude_sip=mpeg4-generic,flv,mpv,h264,vp8

 

[Luca]

We checked the report.
Both logs and traffic dump show there is no media traffic from browser. The call is finished by RTP activity:
View attachment 3357
That's because you've starting a call using REST API /call/startup. In this case, a special media traffic generator must be enabled:

generate_av_for_ua=all

Also, for audio only calls we recommend to exclude all the video codecs:

codecs_exclude_sip=mpeg4-generic,flv,mpv,h264,vp8

Thank you Max

Are the properties you suggested changing in the flashphoner.properties configuration?

Please let me know
Thank you
Luca

 

[Max]

Are the properties you suggested changing in the flashphoner.properties configuration?

Yes, all the settings should be configured in flashphoner.properties file