[Spce-user] Problems with WebRTC

[Nikita Stashkov]

Ok, thank you. So, today it is impossible. I think that config files of the new version of Kamailio are incompatible with the current version.

Regards,
Nikita Stashkov

14 апр. 2014 г., в 16:04, Andreas Granig <agranig at sipwise.com> написал(а):

> Hi,
> 
> On 04/14/2014 12:29 PM, Nikita Stashkov wrote:
>> Some problems using WebRTC client.
>> 1. To work with SIP phones and PSTN we should set Prefer SRTP in user preferences. If it is set, SPCE adds a=crypto:1 AES_CM_128_HMAC_SHA1_80
>> in SDP. So this calls work.
>> 2. But calls between two Web clients will not: 
>> Failed to set remote answer sdp: Session error code: ERROR_CONTENT. Session error description: Cryptos must be empty when DTLS is active..
>> DTLS is enabled by default in new versions of Chrome.
>> 
>> Is it possible to check transport, and if it is ws not to add crypto?
> 
> Version mr3.2.x still uses https://github.com/sipwise/mediaproxy-ng as
> media proxy which does not support DTLS yet.
> 
> Upcoming versions will use our new https://github.com/sipwise/rtpengine,
> which - beside some other things required for full WebRTC bridging like
> rtcp-mux/demux - does support DTLS transcoding.
> 
> If you need to have this particular use case working as of today, you
> need to manually install rtpengine (be aware that this needs the git
> master version of kamailio too, as rtpengine also requires the new
> kamailio rtpengine module).
> 
> Andreas
> 
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