>>> On 3/14/2011 at 04:17 PM, in message <[email protected]>, Michael Scheidell <[email protected]> wrote: > > I SUSPECT.. that sipxbridge is sending a reinvite with G.722 and the > ITSP is not complaining (since its not being sent to the right port/ip > combination) >
Investigate your traces a bit more. SipX does not generate G.722 codec info. Freeswitch can if you have modified it's config outside of sipx. Its your phone that sends the codec preferences. All sipxbridge can do is strip out codecs from the header. So your issue is not due to the port nat per se....but something else. A sipx-trace should quickly find the device sending the G.722 in the header. Note that you don't want to look for the device sending G.722 streams...becuase if you tell sipx to relay a G.722 call it will. SipXbridge (sipxrelay actually) simply relays the RTP back out. You want to find the device sending G.722 in the sip invite. My point is, for the record of all those that search the archives and find long threads about whether or not a port nat works without anything special doing layer 7.....it does work and if nothing else has been verified for basic call functions (call in, call out, transfer, AA, voicemail) by someone who was quite convinced it did not work. -M _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
