[GnomeMeeting-devel-list] SIP, RTP ports and firewall


latest CVS. Using GM with a FWD account. To test GM + Firewall I call 612 fwd (date) or 613 fwd (echo test). Stun is activated. I call from laptop which is behind firewall.

Registration is OK. I call 612: Ok, have sound. Hangup call 613: no sound from there side. Call again 612: no sound.

Close GM, open GM, call 613: echo test is working fine. Hangup call 612: no sound.

So result is whatever number you dial, the first call *after starting GM* is ok, after you loose the sound.

Close GM, tcpdump -i ppp0 udp and host fwd.pulver.com on my gateway

Launch GM, call. I see that used port is 5065 and 5067 and I have audio. Hangup, call again, used port is now 5068 and 5065, etc...

Deduction: RTP port in SIP are 5067 till ...? I Modify with-gconf-editor the ports key from entry gnomemeeting (No ports entry for gnomemeeting-snapshot), it change nothing.

My questions: I think that I could solve my problem (Only one RTP port allowed) if I could modify the RTP port to use. But where is this list? Why GM on second call use the previous RTP port? Does the entry GnomeMeeting in gconf have sense? If yes, why the RTP port defined for H323 are not used?

Could be a good idea to put list of RTP ports in preferences so users could adapt them to there setup (Eg: for all my UA I use the same RTP port range which are my owns)


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