- From: Roman Shpount via GitHub <sysbot+gh@w3.org>
- Date: Fri, 20 Apr 2018 23:20:16 +0000
- To: public-webrtc-logs@w3.org
JSEP is describing how subsequent offers are generated and this language is there in order to support third party call control, where it is unclear if this new offer is going to be used for an existing session update or to connect to a new end point through some sort of signaling agent. In case of SIP this corresponds to the offer in 2XX as described here: https://tools.ietf.org/html/rfc3261#section-14.2. Specifically, in case of SIP, specification is saying that UAS SHOULD include as many media formats and media types that the UA is willing to support. Since in case of WebRTC it is unknown how any offer is going to be used, including all available media formats with currently used format first is the safest option. It allows to use this offer for updating existing session without the codec switch and to establishing a new session through a third party call control. -- GitHub Notification of comment by rshpount Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1838#issuecomment-383245940 using your GitHub account
Received on Friday, 20 April 2018 23:20:20 UTC