Re: [sr-dev] git:master:b4aef50e: rtp_media_server: initial creation of README file

Good question, not available at the moment since there is no call transfer/bridge command yet. We always answer before we start the RTP stream etc. The work is started to bridge call legs and I am hoping to complete it soon. However, bridging early media may be suffisant, in the end "faking" a remote ringing may not be a very good user experience. But maybe we want to play a customized ring back tone. The work in progress is more about activating a few codecs already available as filters for transcoding and SDP negociation on both side, this is one thing that needs to be done in Kamailio since this is not provided by oRTP and Mediastreamer2 and we are using only Kamailio for SIP and SDP. The module will have to provide the missing SDP handling, some of it could be moved in the core or SDPOPs not sure about that.