[Live-devel] Developing VoIP Application

Sebastian Ley sebastian.ley at mmweg.rwth-aachen.de
Sun Feb 6 12:28:00 PST 2005


* Ross Finlayson wrote:

> >I can already get an SDP Message from a caller which I can easily feed to
> >livemedia, following the playCommon example application.
>
> Are you aware of the existing "playSIP" application
> <http://www.live.com/playSIP/>??  It's a SIP client that uses the same base
> code "playCommon.cpp" as "openRTSP".  It sounds like you should just use
> this.

Yes, I found that as well. However the SIP Stack included in licemedia is 
rather limited, so I want to use one which provides higher level 
functionality. Anyway, getting the SDP Message was pretty easy ;-)

> >...tcpdump keeps telling me, that the rtp client wants to access the
> >stream on
> >port 32773:
> >
> >22:57:22.201194 IP coyote.mmweg.32773 > tester.mmweg.32773: UDP, length:
> > 28 22:57:22.201488 IP tester.mmweg > coyote.mmweg: icmp 64: tester.mmweg
> > udp port
> >32773 unreachable
> >
> >Why does it not use 32772 as specified in the SDP Message?
>
> Both ports - 32772 and 32773 - are used.  32772 (the even port) is used for
> RTP; 32773 (the odd port) is used for RTCP.

It seems that kphone does not offer RTCP services, and from the SDP RFC I 
gather that this is correctly annonced by:

m=audio 32772 RTP/AVP 3 97 0

If it advertised RTCP as well the SDP Message should have been

m=audio 32772/2 RTP/AVP 3 97 0

So, is there a possibility to send and recieve RTP packets without RTCP with 
livemedia?

(I don't know if it is standards conform to offer RTP without RTCP over SIP, 
but since kphone works with other softphones as well, I suppose that this is 
not too uncommon.)

Regards,
Sebastian
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