Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains "RTP/AVP" not "RTP/SAVP, and Asterisk rejects it with "We are requesting SRTP for audio, but they responded without it!" Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150312/2a15c775/attachment.html>
Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, "David Cunningham" <dcunningham at voisonics.com> wrote:> Hello, > > Can anyone recommend a particular online WebRTC phone for testing with > Asterisk? > > We tried: > > - JsSIP, but even with the "enable video" checkbox disabled it sends video > options in the INVITE SDP and Asterisk rejects it with "Rejecting secure > video stream without encryption details". > > - sipML5, but it won't register, perhaps something to do with not using > the Asterisk Websocket server (which I don't see an option to choose) > > - Janus, but the INVITE SDP contains "RTP/AVP" not "RTP/SAVP, and Asterisk > rejects it with "We are requesting SRTP for audio, but they responded > without it!" > > Thanks for any suggestions. > > -- > David Cunningham, Voisonics > http://voisonics.com/ > USA: +1 213 221 1092 > UK: +44 (0) 20 3298 1642 > Australia: +61 (0) 2 8063 9019 > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150312/f69d02d8/attachment.html>
Hello David, I'd recommend trying http://sipjs.com/ , it's similar to sipjs but you can choose which kind of media it uses via a configuration object: http://sipjs.com/guides/make-call/ Check out the guides, they are extremely clear and informative: http://sipjs.com/guides/ cheers, Olli 2015-03-12 9:20 GMT+02:00 Mitul Limbani <mitul at enterux.in>:> Sipml5 works. You need to have TLS enabled on asterisk web socket. > > Mitul > On 12-Mar-2015 12:47 PM, "David Cunningham" <dcunningham at voisonics.com> > wrote: > >> Hello, >> >> Can anyone recommend a particular online WebRTC phone for testing with >> Asterisk? >> >> We tried: >> >> - JsSIP, but even with the "enable video" checkbox disabled it sends >> video options in the INVITE SDP and Asterisk rejects it with "Rejecting >> secure video stream without encryption details". >> >> - sipML5, but it won't register, perhaps something to do with not using >> the Asterisk Websocket server (which I don't see an option to choose) >> >> - Janus, but the INVITE SDP contains "RTP/AVP" not "RTP/SAVP, and >> Asterisk rejects it with "We are requesting SRTP for audio, but they >> responded without it!" >> >> Thanks for any suggestions. >> >> -- >> David Cunningham, Voisonics >> http://voisonics.com/ >> USA: +1 213 221 1092 >> UK: +44 (0) 20 3298 1642 >> Australia: +61 (0) 2 8063 9019 >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150312/396c2e1f/attachment.html>