Hi
I am using PJ sip stack to build a sip client on mobile.
Since the PJ sip stack complaints with RFC 3551.
I have gone through some of the samples in PJ . I did not get any idea of how a RTP profile is being implementated . As i am new to C coding i could not understand of how an RTP profile was used in the esablishment of audio calls.
Am i right that PJ sip stack confirms to RFC 3551 to establish audio calls or is it independent of RTP profile.
What iam trying is if there is a RTP profile then use it to extend to video.
Any pointers related to extending video on the RTP side implentation would be greatly appreciated.
Thanks & Regards
Rajeswari.R
Save all your chat conversations. Find them online at http://in.messenger.yahoo.com/webmessengerpromo.php
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20071207/ad362fb6/attachment.html
