[OpenSIPS-Users] Call from non-NATed endpoint to NATedendpointfailed
Leon Li
Leon.Li at aarnet.edu.au
Mon Aug 31 08:10:00 CEST 2009
Ok, I modified the supported codec on the SIP client and the call went through successfully. However, I got one way audio. UA2(callee) with public IP can hear, but UA1(caller) with private IP cannot.
The INVITE msg arrived on UA2 is like with the private IP in SDP. I think this causes the problem because UA2 (public IP) won't know how to get to private ip.
U +0.000413 OPENSIPS_IP:5060 -> UA2_IP:5060
INVITE sip:1001 at UA2_IP:5060 SIP/2.0
Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=i3yf7wxm4qfxlclcjdbbk3tlgzmd.7rv;nat=yes>
Via: SIP/2.0/UDP OPENSIPS_IP;branch=z9hG4bKbf18.6ae7ca36.0
Via: SIP/2.0/UDP 192.168.1.103:5060;received=202.158.213.132;rport=5060;branch=z9hG4bKPjp8-q8skh76pt-ivb9xpujijdh9rna.ep
Max-Forwards: 69
From: "1000" <sip:1000 at OPENSIPS_IP>;tag=i3yf7wxm4qfxlclcjdbbk3tlgzmd.7rv
To: <sip:1001 at OPENSIPS_IP>
Contact: <sip:1000 at 202.158.213.132:5060;transport=UDP>
Call-ID: kslmp2miar4xoard25cwj.leqjm7l9d2
CSeq: 27072 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: SipPhone v3.0/iPhoneOS
Content-Type: application/sdp
Content-Length: 267
P-hint: route(3)|setflag7,forcerport,fix_contact
P-hint: inbound->inbound
P-hint: Route[6]: mediaproxy
v=0
o=- 3460664685 3460664685 IN IP4 192.168.1.103
s=pjmedia
c=IN IP4 192.168.1.103
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
How can I debug and check whether OpenSIPs did NAT/mediaproxy part properly?
Regards,
Leon
-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Saúl Ibarra
Sent: Monday, 31 August 2009 11:22 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Call from non-NATed endpoint to NATedendpointfailed
Are you doing something like this in your opensips configuration script?
if(msg:len>max_len)
{
sl_send_reply("513", "Message too big");
exit;
}
Because that INVITE with the big SDP may be the cause...
Regards,
--
/Saúl
http://www.saghul.net | http://www.sipdoc.net
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