Oh, my confusion is that for some of the SIP phones, even using local phone IP address, the RTP stream for both directions still can be built up. I believe this is due to they have implemented some NAT-traversal mechanism.
Regards/Linda -----Original Message----- From: George Lee [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 17, 2004 7:12 PM To: Linda Xiao Subject: Re: [Sip-implementors] SDP of SIP Invite and RTP stream Hi Linda: In fact, i think that it is due to support NAT/FW traversal for some sip phones. Of course, if using local LAN IP address, incoming RTP media stream can not established fron outside of LAN because of no route. At present, many sip phone vendor even including Cisco have no special solution to NAT/FW issue. Cordially, George Lee China ShenZhen ----- Original Message ----- From: "Linda Xiao" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, August 18, 2004 4:49 AM Subject: [Sip-implementors] SDP of SIP Invite and RTP stream > Hi all, > > > > Supposed that the SIP phones are behind NAT, and the SIP server is on the > internet. For the SDP of SIP Invite, > > > > I have noticed that for some SIP phones, the IP address of both creator and > connection info must be set to the WAN IP address, and then, the RTP stream > for both directions can be built up. If these IP addresses are set up local > phone IP address, then only one direction of RTP stream (phone to server) is > built. > > > > But for some of other phones, even if the IP address for both creator and > connection info is set to local phone IP address, bi-direction of RTP stream > can be built. > > > > Can anyone explain why? > > > > Thanks > > > > Regards/Linda > > _______________________________________________ > Sip-implementors mailing list > [EMAIL PROTECTED] > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > > _______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
