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Issue still occurs if we start the conference after half an hour or so of streaming. It didn't happen when we started within a few minutes of starting the stream. What does this threshhold in the config mean (aac_encoder_sync_drop_threshold=1000000)? Does it only fix the issue if it happens...

Hi, Thanks, that seems to fix it. We will do some more testing. But looking good.

Hi, Can you please contact Twilio about this issue? Since you have a better understanding of how you are capturing this audio and interpreting it. We were under the impression that Twilio was one of the services you supported with SIP to RTMP.

Hi, I emailed log files and a dump containing the rtp packets. The issue occurs around 13:05:35 UTC. You can also find steps to reproduce the issue in the email.

Hi, No luck i'm afraid. The issue still occurs on our wowza server. I tried the new configuration both with and without the earlier proposed rtp_force_synchronization setting. Vlc media player playback failed even before the conference was started. The change caused more synchronization errors...

Hi, I sent an email with the log files. I performed two tests: siptortmp twilio voice app Call using a twilio sip domain to a twiml app that plays music in a loop and stream to the flasphoner rtmp server. We do not experience the issue of audio timecodes resetting. But there are a lot of...

I updated to version: 5.0.2552 and added the line rtp_force_synchronization =true to our properties file. The issue still occurs. It did remove the line ""Audio sync not available, use first packet time!"". The log is now as follows: 08:41:11,886 INFO AudioSession - pool-21-thread-5...

Hi, We are using 'sip to rtmp' to call to a Twilio conference call and stream to a Wowza server. However, as soon as the conference call is started, wowza reports the following: "CupertinoPacketHandler.resetStream[......][11:0]: Timecodes jumped back in time.". The timecode is reset to the...