El Lunes, 15 de Diciembre de 2008, chandan kumar escribió: > Hi , > > Iam testing IP phone which supports video & Audio.Iam facing an issue like > .Iam testing on 2 DSL lines . So End users are on different NAT's. I have > registered both the users to Public SIP servers( freel available servers > for IP calls , using SIPgate).Registration happens.I made call ,call > connects . But there is only Audio & some times no audio & video. > with in India working , But calls from say UK to India are having issues. > > issues are: > > only Audio: from the ethereal captures i can see only Audio Packets > flowing.. > Video freezes:From the captures i can see end points are sending packets to > router WANIP.Iam not sure whether the packets are reaching the endpoint.How > can i check this issue. Are the packets blocked by the router.?Iam using > D-link & Linksys. > For the above issues what could be the possible reason. > > Need clear understanding of STUN in media flow. Does STUN comes in to > picture for RTP flow. How should be the STUN client & STUN server > behavoiur? Is there any relation between STUN & RTP? > Please could any one give brief clarification ,thats enough for me.
Again: This is not the appropiate list for such a question. -- Iñaki Baz Castillo _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
