<div>
<div>Hi,</div><div><br></div><div>So I findally got some time trying WebRTC. I noticed that some clients has no media when using WebRTC. I using the latest master with jssip 3.0. While my chrome(<span style="color: rgb(48, 57, 66); font-family: 'Lucida Grande', sans-serif; font-size: 12px; ">30.0.1599.101)</span> has no problem all the time, other chromes elsewhere seems has no luck. e.g. 31.0.1650.63 and 33.0.1737.0 canary.</div><div><br></div><div>I listed 3 channels below, looks like only my channel has 'AES_CM_128_HMAC_SHA1_80' and other channels has no that and no media. Also I see no auto stun for a long time from the log.</div><div><br></div><div>Before trying more, is any hint it's a Chrome version problem or anything else? the no audio stun for a long time log has anything related to the client or only related to server side? Any configurable params? Thanks.</div><div><br></div><div><br></div><div><br></div><div><br></div><div>L16,48000,768000,opus,48000,0,srtp:sdes:AES_CM_128_HMAC_SHA1_80,webRTC,1000@115.29.169.114,,ACTIVE,,,,,,</div><div>L16,48000,768000,opus,48000,0,,webRTC,1000@115.29.169.114,,ACTIVE,,,,,,</div><div>L16,48000,768000,opus,48000,0,,webRTC,1000@115.29.169.114,,ACTIVE,,,,,,</div><div><br></div><div>3 total.</div><div><br></div><div>2013-12-13 19:18:24.459945 [ERR] switch_rtp.c:718 No audio stun for a long time!</div><div>2013-12-13 19:18:35.479939 [ERR] switch_rtp.c:718 No audio stun for a long time!</div>
</div>
<div><div><br></div><div>-- </div><div>Seven Du</div><div><span style="font-size: 10pt; ">http://www.freeswitch.org.cn</span></div><div>http://about.me/dujinfang</div><div><div>http://www.dujinfang.com</div></div><div><br></div><div>Sent with <a href="http://www.sparrowmailapp.com/?sig">Sparrow</a></div><div><br></div></div>