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[dcunningham at voisoni...]

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
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[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@voisonics.com (dcunningham@voisonics.com)> wrote:

[ohjelmistoarkkitehti a...]

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@enterux.in (mitul@enterux.in)>: