[Spce-user] Problems with WebRTC

[Andreas Granig]

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