<div dir="ltr">Hi Lorenzo,<div><br></div><div>Not trying to hijack the thread, but Andrea appears to be away for the next few days, and I'm experiencing the same exact problem with the audio issues and want to help move us both forward. So I can answer the questions below to get you more data. Please see below:<br>
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<div class="im"><span style="color:rgb(34,34,34)">[May 28 11:17:40] DEBUG[16613][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer tic</span><br></div><div class="im"><br>
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That said, I'm not sure what the cause of the issue could be. What ptime are the two peers using?<br>
What is the transcoding path<br>
as displayed on the Asterisk console? Does the same happen in other scenarios as well, e.g., the browser interacting with another<br>
browser, or a softphone using a different codec (e.g., speex) at different rates (e.g., 16kHz vs 8kHz)?<br>
</blockquote>
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now I'm out of office I will be able to provide this info<br>
only on Thursaday.<div class="im"><br></div></blockquote><div><br></div></div><div style>I assume the ptime is this maxptime value we see in the SDP from the Chrome to Asterisk call leg:</div><div><br></div><div>a=rtpmap:111 opus/48000/2</div>
<div>a=maxptime:60</div><div>a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0</div><div class="gmail_extra"><br></div><div class="gmail_extra" style>If that's not the ptime, please let me know where I might find that and I'll get it for you.</div>
<div class="gmail_extra"><br></div><div class="gmail_extra" style>And here you can see the call leg from the Chrome to Asterisk portion of the call is opus, with the other leg using ulaw. I've tried with g729 on the Bandwidth to Asterisk leg and still hear the same underwater-like call audio that Andrea describes.</div>
<div class="gmail_extra"><br></div><div class="gmail_extra">ip-10-188-135-200*CLI> sip show channels</div><div class="gmail_extra">Peer User/ANR Call ID Format Hold Last Message Expiry Peer </div>
<div class="gmail_extra">54.X.X.X 1000 55srb9arjvh7rnv (opus) No Rx: ACK ws-Opensip</div><div class="gmail_extra">67.231.X.X +15036XXXXXX 48c000dd54ba525 (ulaw) No Tx: ACK BANDWIDTH </div>
<div class="gmail_extra">2 active SIP dialogs</div><div class="gmail_extra">ip-10-188-135-200*CLI> </div><div><br></div><div><br></div><div style>Also, here is the output from opus set debug huge:</div><div style><br>
</div>
<div style><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 18 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 20 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 77 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 19 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 67 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 17 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 71 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 21 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 21 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 23 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 69 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 19 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 21 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 74 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 23 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 74 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 21 bytes</div><div>[Opus] [Decoder #12 (8000)] 960 samples, 75 bytes</div><div>[Opus] [Decoder #12 (8000)] >> Got 160 samples, 320 bytes</div>
<div>[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes</div><div>[Opus] [Encoder #9 (8000)] >> Got 960 samples, 25 bytes</div></div><div class="gmail_extra"><br></div><div class="gmail_extra"><br></div>Unlike Andrea, who applied the patch to 11.3.0, I successfully applied the patch to Asterisk 11.4.0, but the issues we're both experiencing appear to be the same. If that's not the case, I can start a new thread. Hope this helps! :)<br clear="all">
<div><br></div>-- <br>James Mortensen<br>Project Manager, VoiceCurve, Inc.<br>866-707-4590<br><a href="mailto:james.mortensen@voicecurve.com" target="_blank">james.mortensen@voicecurve.com</a><br>
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