[asterisk-users] WebRTC demo phones

[Olli Heiskanen]

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
>>
>> --
>> _____________________________________________________________________
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>
> --
> _____________________________________________________________________
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