[Spce-user] Problems with WebRTC

Nikita Stashkov snl at sipmobile.org
Mon Apr 14 06:29:02 EDT 2014


Hello,

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?

Regards,
Nikita Stashkov


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