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Hi All,</div>
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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.</div>
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<img style="width: 908.693px; height: 806px; max-width: initial;" class="w-1637 h-1452" width="908" height="806" size="218830" contenttype="image/png" data-outlook-trace="F:1|T:1" src="cid:cfdedbcf-5048-4f07-8fe1-1c761ffdfee6"><br>
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<img style="width: 1143.03px; height: 432px; max-width: initial;" class="w-2157 h-815" width="1143" height="432" size="173658" contenttype="image/png" data-outlook-trace="F:1|T:1" src="cid:768e2f51-801b-4411-aade-a77cecad6d45"><br>
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We've tried with and with-out the following, but there's no change.
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<li>bridge_generate_comfort_noise</li><li>send_silence_when_idle</li><li>suppress_cng</li><li style="font-family: Calibri, Arial, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);" class="elementToProof">
suppress-cng (SIP profile param)<br>
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Has anyone come across this before, and is there a solution?</div>
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Thanks,</div>
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Shaun</div>
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