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``` [deleted comment] ```
``` I have the same issue with ending call from callee when on wss and works ok with ws. (using chrome 39, asterisk 13). does anyone have solution? ``` Original...
``` https://groups.google.com/group/doubango/browse_thread/thread/679ac2c7dc5428f6 ``` Original comment by `[email protected]` on 24 May 2013 at 1:44 - Changed state: **Accepted**
``` In this case can there be a problem in message integrity part of stun responses (for example wireshark packet no:13773, 13775)? Thanks ``` Original comment by `[email protected]` on 31...
``` Some problem here. Webrtc2SIP r99 Doubango r904 On Ubuntu Server 12.04 Simpl5 latest ``` Original comment by `[email protected]` on 25 Jul 2013 at 10:45
``` Enable RTCWeb Breaker. ``` Original comment by `[email protected]` on 23 Jul 2013 at 6:19
Original comment by `[email protected]` on 7 May 2013 at 6:45 - Changed title: **Calling PSTN create resampler (opus->g711)**
``` https://code.google.com/p/webrtc2sip/wiki/FAQ#How_to_report_issue? ``` Original comment by `[email protected]` on 1 Aug 2013 at 7:32
``` The problem was in ptime as Gökhan Barış Aker said. Legacy SIP sends rtp data every 30 ms and when I've change it to 20 ms, everything become ok....
``` what's your doubango version? ``` Original comment by `[email protected]` on 5 Mar 2013 at 5:26