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.