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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