Hello, I was able to use webrtc2sip and connect audio calls in g729 passthrough and ulaw modes over a callus webpage js.
However not tested Video. and it worked good even on AST 1.8.XX Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: [email protected] DID: +91-22-71967196 Cell: +91-9820332422 On Mon, Apr 14, 2014 at 2:26 PM, Johan Wilfer <[email protected]> wrote: > Hi, > > I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + > opus/vb8 codec patch. This is interesting technology and I try to find out > how to connect all the moving parts. > > Firefox: > Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't > matter. > WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without > encryption details: audio 35684 RTP/SAVPF 109 0 8 101 > --> Asterisk sends "SIP/2.0 488 Not acceptable here" > > Chrome: > I've tried both sipml5 and jssip softphones and they both work. Even video > + confbridge works with some minor quirks (lost connections sometimes, I > guess plain old nat issues). > Just relaying audio+video with confbridge to a handful of participants > seems to use quite a bit of cpu thought. > > Screen-share: > This works, but Confbridge is not very happy about a channel with video > (vp8) and not audio and is printing this 80 times a second: > > WARNING[8919][C-00000000] channel.c: Unable to find a codec translation > path from (vp8) to (slin) > WARNING[8919][C-00000000] chan_sip.c: Asked to transmit frame type slin, > while native formats is (vp8) read/write = unknown/unknown > WARNING[8919][C-00000000] channel.c: Don't know any of (vp8) formats > > > How do you think about adding webrtc to a existing Asterisk/Kamailio > environment? Do you use kamailio (websockets) as a front, a dedicated > webrtc asterisk or something like webrtc2sip? > > How do you use / plan to implement webrtc in your environment? > > Any feedback is welcome. Thanks! > > -- > Johan Wilfer > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
