Hello,
some user agents are configured to send rtp to the address and port from
where they receive rtp (so called co-media support). It is an
explanation that can be there. I see no other reason, if the media IP is
not changed.
Cheers,
Daniel
On 10/17/07 15:27, Iñaki Baz Castillo wrote:
Hi, user_A with STUN calls to user_B behind NAT with
no STUN.
onreply_route[1] {
if (nat_uac_test("1"))
fix_nated_contact();
if (isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]")
force_rtp_proxy("l");
}
In the initial request bflag(6) is up because "location" of user_B so RtpProxy
is used in the INVITE and 200-OK.
But in re-INVITE bflag(6) is down and is not applied force_rtp_proxy("l"); in
the 200-OK.
In fact I debug the SIP trace in the re-INVITE in OpenSer:
----------------------------------------------------------------------
# user_A -> OpenSer
INVITE sip:user_B@112.121.235.28:5061 SIP/2.0
c=IN IP4 212.121.235.18
# OpenSer -> user_B
INVITE sip:806@212.121.235.18:5061 SIP/2.0
c=IN IP4 80.94.0.110 <---- RtpProxy applied (OK)
# user_B -> OpenSer
SIP/2.0 200 OK
c=IN IP4 192.168.1.106 [*1]
*1: Needs RtpProxy but bflag(6) is down so not applied.
# OpenSer -> user_A
SIP/2.0 200 OK
c=IN IP4 192.168.1.106 <-- RtpProxy NO applied
----------------------------------------------------------------------
But I can confirm that the audio works after re-INVITE in both directions!!!
¿?
I've debugged with tcpdump, the RTP is send to RtpProxy from user_A, how is
possible??
Thanks.