Linda,
If you have a SIP phone sitting behind a firewall, then it can be made to work in many ways like: * The firewall is SIP aware, and it maps all the private IP addresses in the SIP SDP to internet IP addresses. This firewall is then effectively providing NAT for SIP traffic too. * There is a configuration option with in the SIP Phone to utilize the pin holes opened by the Firewall for SIP media traffic. For example, a SIP phone in a private lan maybe allocated following tuple <internet-address-1, port-a> then this port/IP address may be used by the SIP Phone in its SDP. When the firewall, sees traffic coming on this port it always forwards it to a default IP address. * Another option is to have a B2BUA, which is interfacing with both private network and the internet. This B2BUA then relays media stream between private LAN and WAN. This is a simple method implemented in a lot of devices including one from yours truly. It has the added advantage that all available SIP phones work right off. [PS : please dont flame me for this ;-) Linda, Could you tell me in a private e-mail, the SIP phones you are using and I may be able to help you better.] Arun Punj ViPr Software Engineer Broadband Routing & Switching Marconi 3000 Marconi Drive Warrendale, PA 15086 Phone: 724-742-7583 [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > Linda Xiao > Sent: Wednesday, August 18, 2004 7:28 PM > To: 'George Lee' > Cc: '[EMAIL PROTECTED]' > Subject: RE: [Sip-implementors] SDP of SIP Invite and RTP stream > > > 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 > _______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
