[OpenSIPS-Devel] [ opensips-Bugs-3150707 ] nathelper adding extra CRLF before inserting "a=nortpproxy"

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Tue Jan 4 16:40:55 CET 2011


Bugs item #3150707, was opened at 2011-01-04 02:29
Message generated for change (Comment added) made by bogdan_iancu
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Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: Robert Smith (denodaeus)
>Assigned to: Bogdan-Andrei Iancu (bogdan_iancu)
Summary: nathelper adding extra CRLF before inserting "a=nortpproxy"

Initial Comment:
It seems like there are a few reports of this happening, but I've got a case where I'm using nathelper to rtpproxy_offer and I get the below situation.  Note the extra CRLF after the G711U and before the nortpproxy:yes:

This is current opensips trunk:

U 2011/01/03 22:20:52.192745 4.2.2.210:5060 -> 4.2.2.150:5060
INVITE sip:9001 at sip36660.aaa.mysipserver.com SIP/2.0.
Record-Route: <sip:4.2.2.210;lr=on;nat=yes>.
Via: SIP/2.0/UDP 4.2.2.210;branch=z9hG4bK9077.d4132cd2.0.
Via: SIP/2.0/UDP 192.168.30.18:5066;received=192.168.30.18;rport=5066;branch=z9hG4bK1737023440.
From: <sip:sipuser at sip-36660.bbb.mysipserver.com>;tag=674081069.
To: <sip:9001 at sip-36660.bbb.mysipserver.com>.
Call-ID: 2752934103.
CSeq: 20 INVITE.
Contact: <sip:sipuser at 192.168.30.18:5066>.
Content-Type: application/sdp.
Max-Forwards: 69.
User-Agent: eXosip/3.1.0.
Subject: click2dial call.
Expires: 120.
Content-Length:   225.
.
v=0.
o=click2dial 0 0 IN IP4 192.168.30.18.
s=click2dial call.
c=IN IP4 192.168.30.18.
t=0 0.
m=audio 8000 RTP/AVP 0 8 18 3 4 97 98.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:97 ilbc/8000.
a=rtpmap:98 speex/8000.


U 2011/01/03 22:20:52.211277 172.16.30.150:5060 -> 10.2.1.80:5060
INVITE sip:9001 at sip36660.aaa.mysipserver.com SIP/2.0.
Record-Route: <sip:172.16.30.150;r2=on;lr=on;did=a82.01335004;destination=10.2.1.80>.
Record-Route: <sip:4.2.2.150;r2=on;lr=on;did=a82.01335004;destination=10.2.1.80>.
Record-Route: <sip:4.2.2.210;lr=on;nat=yes>.
Via: SIP/2.0/UDP 172.16.30.150;branch=z9hG4bK9077.b2ed18b1.0.
Via: SIP/2.0/UDP 4.2.2.210;branch=z9hG4bK9077.d4132cd2.0.
Via: SIP/2.0/UDP 192.168.30.18:5066;received=192.168.30.18;rport=5066;branch=z9hG4bK1737023440.
From: <sip:sipuser at sip-36660.bbb.mysipserver.com>;tag=674081069.
To: <sip:9001 at sip-36660.bbb.mysipserver.com>.
Call-ID: 2752934103.
CSeq: 20 INVITE.
Contact: <sip:sipuser at 192.168.30.18:5066>.
Content-Type: application/sdp.
Max-Forwards: 68.
User-Agent: eXosip/3.1.0.
Subject: click2dial call.
Expires: 120.
Content-Length: 166.
.
v=0.
o=click2dial 0 0 IN IP4 192.168.30.18.
s=click2dial call.
c=IN IP4 172.16.30.162.
t=0 0.
m=audio 26430 RTP/AVP 0 8 3 4 .
a=rtpmap:0 PCMU/8000.
.
a=nortpproxy:yes

----------------------------------------------------------------------

>Comment By: Bogdan-Andrei Iancu (bogdan_iancu)
Date: 2011-01-04 17:40

Message:
Robert, do you still see this problem when not using codec ops (but only
rtpptoxy functions) ?

Also, what version/revision of opensips are you working with ?

Regards,
Bogdan

----------------------------------------------------------------------

Comment By: Robert Smith (denodaeus)
Date: 2011-01-04 02:33

Message:
Note, the codecs here are manipulated with textops to remove several values
(with codec_delete_except_re("")) BEFORE invoking rtpproxy_offer().

----------------------------------------------------------------------

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