[asterisk-users] WebRTC demo phones
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
Re: [asterisk-users] WebRTC demo phones
Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, David Cunningham dcunning...@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
Re: [asterisk-users] WebRTC demo phones
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 mi...@enterux.in: Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, David Cunningham dcunning...@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 -- _ -- 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
