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Dear All, I use sipml5 to register two users from browser and the two clients are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in audio

....5.49 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiBw81ooU7ybSRbRqr8TOqWkMPQRdkMXo;rport From: "John Doe (101)"<sip:1060 at 10.10.5.49>;tag=heMv1HvlT7DeQxPxuqcq To: <sip:204 at 10.10.5.49> Contact: "John Doe (101)"<sip:1060 at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=0b2413e6f3c96a0517b4413a6f6ce7ae;+g.oma.sip-im;+sip.ice;language="en,fr,it" Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc CSeq: 38718 INVITE Content-Type: application/sdp Content-Length: 1827 Max-Forwards: 70 User-Agent: IM-client/O...

...d;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;rport [Aug 9 22:15:50] From: "77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl [Aug 9 22:15:50] To: <sip:419 at 178.18.90.230> [Aug 9 22:15:50] Contact: "77"<sips:770000wrtc at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" [Aug 9 22:15:50] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [Aug 9 22:15:50] CSeq: 21553 INVITE [Aug 9 22:15:50] Content-Type: application/sdp [Aug 9 22:15:50] Content-Length: 1815 [Aug 9 22:15:50] Max-Forwa...

I too would like to know the status of the opus IP holders. From what I've seen on the CELT and opus lists, it seems that the CELT part of opus is covered by patents from Xiph and Broadcom, and the SILK part of opus is covered by the rest. All of the IP holders except Qualcomm and Huawei have released their patents to the public as far as opus is concerned, and the general consensus from

...2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;rport [Aug 11 15:53:47] From: <sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ [Aug 11 15:53:47] To: <sip:419 at 178.18.90.230> [Aug 11 15:53:47] Contact: <sips:770000wrtc at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" [Aug 11 15:53:47] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec [Aug 11 15:53:47] CSeq: 58874 INVITE [Aug 11 15:53:47] Content-Type: application/sdp [Aug 11 15:53:47] Content-Length: 2301 [Aug 11 15:53:47] Max-Forwa...

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

...at ipbx SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;rport From: "cervenka"<sip:vr1a882 at vhXXX.example.com>;tag=RDmpGm2Mubc5xQQ8NMli To: <sip:887 at ipbx> Contact: "cervenka"<sips:vr1a882 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" Call-ID: cf2990ba-3f12-3d9e-adb6-52889c414ed3 CSeq: 41942 INVITE Content-Type: application/sdp Content-Length: 1250 Max-Forwards: 70 Authorization: Digest username="vr1a882",realm="pbx",nonce=&q...