[Live-devel] Developing VoIP Application

[Ross Finlayson]

>I am developing a voice oder IP application based on SIP. While the SIP part
>was pretty easy so far, I am having trouble with the negotiated RTP sessions.
>I want to use livemedia for that part as it seem to offer most of the things
>I need.
>
>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.


>...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.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>