Here is use case for using source ip/port. Polycom IP Phones dont support rport and requires you to configure NAT IP / PORT on the phone and punching a hole for every ip phone is not a good idea I guess the only alternative is to let the NAT send all packets to the call manager (open nat one way) and use source ip and source port for routing.
On Mon, Sep 29, 2008 at 9:48 AM, Iñaki Baz Castillo <[EMAIL PROTECTED]> wrote: > 2008/9/29, M. Ranganathan <[EMAIL PROTECTED]>: > > That has NOTHING to do with rport. XMPP is a whole lot simpler than > > SIP. Try doing call transfer, redundancy, fail over, load balancing > > etc with xmpp. XMPP is good for IM and simple point to point voice > > calls (jingle). SIP is simply over engineered for point to point > > instant messaging and presence - like shooting a fly with a cannon. > > The only reason to use SIP for such things as presence is if you are > > already using it for multimedia. > > Completely agree here. > > IMHO the big error comes from the origin of SIP design, that assumes > there is no NAT and defines replying by looking at the received "Via" > header instead of replying directly to the real soure IP:**port**. > I really wonder why. It would be much easier if replies are sent, by > default, to the source IP:port when the transport protocol is UDP (SIP > TCP already ahs that behaviour by design). > > Regards. > > > > > -- > Iñaki Baz Castillo > <[EMAIL PROTECTED]> > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
