5 Aug
2004
5 Aug
'04
8:35 a.m.
Hi Richard,
I'm experiencing similar problems..
NATPhone -------------- SER ------------------- Public phone
the Ser-server has a public ip-address. They both register fine but
when I make a call from the natphone to the publicPhone the public
phone can hear my voice but I can't hear the publicPhone.
INVITE from the log file:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 10.0.0.2:5070;received=213.219.137.171;rport=5070^M
Supported: replaces^M
User-Agent: SIP201 (lp201sip.100a)^M
Contact: <sip:joachim@212.71.17.178:5070>^M
Record-Route:
<sip:1@212.71.0.60:5070;lr;ftag=a000002-13ce-41123427-943ea0-4d15>^M
From: <sip:bart@ser.edpnet.net:5070>
;tag=a000002-13ce-41123427-943ea0-4d15^M
To: <sip:1@ser.edpnet.net:5070;user=phone>
;tag=d44711b2-13ce-1135c2-433a03fc-1d24^M
Call-ID: 522c24-a000002-13ce-41123427-943e9b-28ff(a)ser.edpnet.net^M
CSeq: 1 INVITE^M
Content-Type: application/sdp^M
Content-Length:158^M
^M
v=0^M
o=SIP201 12367 0 IN IP4 212.71.17.178^M
s=Audio Session^M
i=Audio Session^M
c=IN IP4 212.71.17.178^M
t=0 0^M
m=audio 16384 RTP/AVP 4^M
a=rtpmap:4 G723/8000/1^M
When I make a Call from the PublicPhone to the natPhone We both don't
hear a thing and the INVITE is like this:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 212.71.17.178:5070^M
Supported: replaces^M
User-Agent: SIP201 (lp201sip.100a)^M
Contact: <sip:bart@10.0.0.2:5070>^M
Record-Route:
<sip:2@212.71.0.60:5070;lr;ftag=d44711b2-13ce-113882-4344c26a-146f>^M
From: <sip:joachim@ser.edpnet.net:5070>
;tag=d44711b2-13ce-113882-4344c26a-146f^M
To: <sip:2@ser.edpnet.net:5070;user=phone>
;tag=a000002-13ce-411236e9-9f0123-31fd^M
Call-ID: 522dc4-d44711b2-13ce-113882-4344c265-649(a)ser.edpnet.net^M
CSeq: 1 INVITE^M
Content-Type: application/sdp^M
Content-Length:148^M
^M
v=0^M
o=SIP201 12367 0 IN IP4 10.0.0.2^M
s=Audio Session^M
i=Audio Session^M
c=IN IP4 10.0.0.2^M
t=0 0^M
m=audio 16384 RTP/AVP 4^M
a=rtpmap:4 G723/8000/1^M
Is this because there's a private address (10.0.0.2) in the sipmessage?
thanks,
Bart
-----Original Message-----
From: Richard [mailto:mypop3mail@yahoo.com]
Sent: woensdag 4 augustus 2004 12:15
To: serusers(a)lists.iptel.org
Subject: [Serusers] rtpproxy with parallel forking
Hi,
When I have one phone behind NAT, rtpproxy works well.
However if a call goes to two phones (one behind NAT and one
not) and ser (or more precisely cpl-c) does parallel forking,
the call can go through but there is no voice on either
direction. I have a modparam("cpl-c", "proxy_route", 2) and
route(2) calls force_rtp_proxy().
Has anyone tried this before? Is rtpproxy capable to handle
multiple forked calls?
Thanks,
Richard
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