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

[Henning Westerholt]

Am Mittwoch, 14. November 2018, 07:00:08 CET schrieb 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.

Hi Julien,

there is SDP parser functionality in the core (core/parser/sdp/sdp.h), if you 
need more functionality for parsing this should be probably extended there. 
The sdpops module then provides SDP handling function to use in the cfg file, 
using the core. Maybe it would be sufficient to use existing functionality 
here, or change/add some new functions in this module.

Best regards,

Henning


-- 
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://skalatan.de/services
Kamailio security assessment - https://skalatan.de/de/assessment