Google Code Exporter

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``` Any chance to get this implemented ``` Original comment by `[email protected]` on 19 May 2013 at 5:09

``` ANy update on this? ``` Original comment by `[email protected]` on 19 Jun 2013 at 1:44

``` Using SIP RTP CoS mark 5 [Apr 7 18:44:14] WARNING[11008][C-0000001f]: chan_sip.c:10620 process_sdp: Rejecting secure audio stream without encryption details: audio 11358 RTP/SAVPF 111 103 104 0 8 106 105...

``` [deleted comment] ```

``` I also having this problem . When using Google Chrome ( v35.0.1916.114 m), I see it in SIP SDP: ( I have no glue where address 10.0.2.15 comes from,...

``` https://code.google.com/p/webrtc2sip/wiki/FAQ#How_to_report_issue? ``` Original comment by `[email protected]` on 2 Jun 2014 at 2:15

``` I think I found the reason for this: On an incoming call, the sipml JS stack immediately starts the RTCPeerconnection. When the user waits a few seconds, the peerconnection...

``` Hi, have you solved this problem? I have the exact issue right now. Thank you! ``` Original comment by `[email protected]` on 6 Jan 2015 at 7:28

``` ^^^^^^ The webrtc2sip logs were taken when chrome was connected ``` Original comment by `[email protected]` on 16 Apr 2015 at 3:11

Original comment by `[email protected]` on 9 Jan 2014 at 5:40 Attachments: - [webrtc2sip-log](https://storage.googleapis.com/google-code-attachments/webrtc2sip/issue-148/comment-1/webrtc2sip-log)