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

[Julien Chavanton]

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.



On Tue, Nov 13, 2018, 21:19 Juha Heinanen <jh at tutpro.com wrote:

> One more question:  at this stage, rtp_media_server is not able to play
> early audio?
>
> -- Juha
>
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