[OpenSIPS-Users] Call from non-NATed endpoint to NATedendpointfailed

[Leon Li]

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

_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users