[SR-Users] SIP - WebRTC gateway

Hello, I'm figuring out the best approach to deploy a bridge between Websocket\Webrtc and SIP\rtp. Can Kamailio (+mediaproxy-ng or something else) operate as a full Webrtc\SIP gateway (signaling, audio or video transcoding, ICE and so on)? Some months ago I found the architecture described here

[SR-Users] retrieve INVITE transaction in CANCEL

Hello, is it possible to retrieve the info related to the INVITE transaction while the CANCEL is processed? In particular, when the script is handling the CANCEL, how can I get the avp values written in the corresponding INVITE transaction? Thanks, Daniele **