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@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@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@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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