[OpenSER-Users] Rewrite SDP for music of hold

Christian Schlatter cs at unc.edu
Tue Jul 10 20:03:53 CEST 2007


Iñaki Baz Castillo wrote:
> Hi, I'm starting using OpenSer with Asterisk.
> 
> In calls passing through Asterisk the caller or callee get music on hold of 
> Asterisk when is put on hold. But of course, direct calls between OpenSer 
> users don't get this music.
> 
> I'm thinking in the posibility of detecting "on hold" reinvites (maybe reading 
> the "a=sendonly" of SDP body) and rewriting the SDP IP contact to a music on 
> hold server.

You would also have to tell the music server where to send the rtp 
stream. And you'd have to keep dialog state on the proxy in order to 
capture the off-hold INVITE and tell the music server to stop sending media.

This all leads to a sip proxy acting as a B2BUA/3PCC server which needs 
a lot of resources and is difficult to implement. I don't say it is 
impossible but adding this functionality to openser would be quite a task.

Unfortunately even the IETF has no clear message on music on hold, no 
wonder sip phones normally do not support it in a standards compliant 
way. There is a draft called "SIP Service Examples" that already has 12 
revisions and that includes an example on how music on hold should be 
implemented.

http://www1.tools.ietf.org/html/draft-ietf-sipping-service-examples-12#section-2.3

Previous revisions of this draft followed a 3PCC model for music on hold 
(http://tech-invite.com/Ti-sip-service-3.html), whereas revision 12 now 
implements music on hold with a call transfer.

One can only hope that we will see products implementing these procedures.

Pingtel seems to offer a music-on-hold server for the 3PCC model: 
http://interop.pingtel.com/#moh

And SNOM phones seem to work fine together with the pingtel/sipX 
solution: 
http://sipx-wiki.calivia.com/index.php/HowTo_configure_SNOM_SIP_phone_with_sipX#Music_On_Hold_.28MoH.29


Christian


> I have no idea if this is possible. The first issue is how to set up a RTP 
> music streamer for pointing there the SDP c header. Does "this" RTP music 
> streamer exist? of course Asterisk is not a solution since I don't want to 
> manage INVITE's, just rewrite the SDP.
> 
> Second issue is: could be rewriting the SDP always sucessfull? what about 
> cases with more than one RTP session as audio+video?
> 
> So, is there any solution for offering music on hold with OpenSer?
> 
> Thanks a lot.
> 
> 




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