[OpenSER-Users] One way audio calling a -> b, two way audio calling b -> a with two phones behind the same NAT
Hi,
I am having some trouble with nat traversal. Depending on which phone is originating the call, it may either have one way audio, two way, or even neither.
Here is the matrix I have tested so far. Hardphone to hard phone works fine between two spa942 or either of the spa942 phones and the budgetone, the problems come into the picture when soft phones are used.
caller callee RTP direction budgetone linphone neither budgetone spa942 both budgetone xlite both linphone budgetone one way linphone spa942 one way linphone xlite neither spa942 budgetone both spa942 linphone neither spa942 xlite one way xlite budgetone both xlite linphone both xlite spa942 both
Openser.cfg http://pastebin.ca/946561
El Monday 17 March 2008 21:00:09 Mike Fedyk escribió:
Hi,
I am having some trouble with nat traversal. Depending on which phone is originating the call, it may either have one way audio, two way, or even neither.
Capture the sip trace and examinate the addresses in SDP of INVITE and 200 Ok (and maybe ACK). Also do a tcpdump to know where is going the rtp.
Thanks, I had a misconfigured arp-proxy setup on my testing setup.
Mike
-----Original Message----- From: users-bounces@lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf Of Iñaki Baz Castillo Sent: Tuesday, March 18, 2008 1:37 AM To: users@lists.openser.org Subject: Re: [OpenSER-Users] One way audio calling a -> b,two way audio calling b -> a with two phones behind the same NAT
El Monday 17 March 2008 21:00:09 Mike Fedyk escribió:
Hi,
I am having some trouble with nat traversal. Depending on which phone is originating the call, it may either have one way audio, two way, or even neither.
Capture the sip trace and examinate the addresses in SDP of INVITE and 200 Ok (and maybe ACK). Also do a tcpdump to know where is going the rtp.
-
Iñaki Baz Castillo -
Mike Fedyk