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[Freeswitch-users] Recording calls causes FS generated RTP packets with different timestamp on missed packets


 
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shaun at sysconfig.cloud
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PostPosted: Sun Jul 10, 2022 11:34 pm    Post subject: [Freeswitch-users] Recording calls causes FS generated RTP p Reply with quote

Hi All,


We're having issues when recording calls, using either record_session or uuid_record. FreeSWITCH is generating RTP packets to fill in the gaps when there's missed RTP packets (packet loss), but the FreeSWITCH generated RTP packets use a different timestamp. This causes jitter and periods of silence on active calls. The only solution seems to be enabling RTP rewrite timestamps, but this causes audio delay\sync issues.












We've tried with and with-out the following, but there's no change.
  • bridge_generate_comfort_noise
  • send_silence_when_idle
  • suppress_cng
  • suppress-cng (SIP profile param)




Has anyone come across this before, and is there a solution?


Thanks,
Shaun
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shaun at sysconfig.cloud
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PostPosted: Mon Jul 11, 2022 11:34 am    Post subject: [Freeswitch-users] Recording calls causes FS generated RTP p Reply with quote

I've re-tested using version 1.10.1, FreeSWITCH still generates RTP packets to fill in the gaps when there's missed RTP packets, but the RTP timestamps are now correct which resolves the issue.


I'll raise the issue on GitHub, as this appears to be a bug.




From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Shaun Stokes <shaun@sysconfig.cloud>
Sent: 09 July 2022 13:18
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: [Freeswitch-users] Recording calls causes FS generated RTP packets with different timestamp on missed packets

Hi All,


We're having issues when recording calls, using either record_session or uuid_record. FreeSWITCH is generating RTP packets to fill in the gaps when there's missed RTP packets (packet loss), but the FreeSWITCH generated RTP packets use a different timestamp. This causes jitter and periods of silence on active calls. The only solution seems to be enabling RTP rewrite timestamps, but this causes audio delay\sync issues.












We've tried with and with-out the following, but there's no change.
  • bridge_generate_comfort_noise
  • send_silence_when_idle
  • suppress_cng
  • suppress-cng (SIP profile param)




Has anyone come across this before, and is there a solution?


Thanks,
Shaun
Back to top
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