[OpenSIPS-Devel] [ opensips-Bugs-3415264 ] recieved=x.x.x.x added to wrong via header
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Thu Sep 29 15:59:28 CEST 2011
Bugs item #3415264, was opened at 2011-09-29 10:39
Message generated for change (Comment added) made by mtryfoss
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Category: None
Group: None
Status: Open
Resolution: Invalid
Priority: 5
Private: No
Submitted By: Morten Tryfoss (mtryfoss)
Assigned to: Bogdan-Andrei Iancu (bogdan_iancu)
Summary: recieved=x.x.x.x added to wrong via header
Initial Comment:
Hi!
It seems like 'recived' is added to the wrong header when the dialog contains multiple comma separated Via-headers. It's just appended to the end of the line.
Example:
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK0440914511a92062ea12e98d4, SIP/2.0/TCP 10.3.8.19:5062;branch=z9hG4bKaad385d13b702b61ee38b3118;rport=4108;received=80.239.13.209
The correct would be:
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK0440914511a92062ea12e98d4;rport=4108;received=80.239.13.209, SIP/2.0/TCP 10.3.8.19:5062;branch=z9hG4bKaad385d13b702b61ee38b3118
It's working ok when you have one Via pr line:
Via: SIP/2.0/UDP 172.50.1.8;branch=z9hG4bK6257.ce0c3755.0;i=1;received=85.19.212.117
Via: SIP/2.0/TCP 172.50.1.9:5060;received=172.50.1.9;branch=z9hG4bK397aee9f;rport=39350
Regards,
Morten Tryfoss
----------------------------------------------------------------------
>Comment By: Morten Tryfoss (mtryfoss)
Date: 2011-09-29 15:59
Message:
The call disconnects on the caller-side as soon as it received the 183
Session Progress.
Complete trace:
INVITE sip:48485858 at 85.19.212.36;transport=tcp SIP/2.0
To: <sip:48485858 at 10.3.8.20:5060>
From: "Unknown" <sip:38531300 at dialer2.krs>;tag=65480
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK117c679a7a564bb8609b56e88,
SIP/2.0/TCP 10.3.8.19:5062;branch=z9hG4bK54b1106da2bc4a3d07f6ace9a
Call-ID: 271c333045c13cf007cb1519b5c09c6f at 10.3.8.19
CSeq: 1 INVITE
Contact: <sip:38531300 at 10.3.8.19:5062;transport=tcp>
Max-Forwards: 69
x-inin-crn: 1001090768;loc=Kristiansand;ms=NECIC4
Supported: join, replaces
User-Agent: ININ-TsServer/3.11.11.12100
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER,
SUBSCRIBE
Accept: application/sdp
Accept-Encoding: identity
Content-Type: application/sdp
Content-Length: 197
Record-Route: <sip:10.3.8.20:5060;lr;transport=tcp>
v=0
o=ININ 2813816323 2813816324 IN IP4 10.3.8.19
s=Interaction
c=IN IP4 10.3.8.20
t=0 0
m=audio 28804 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 100 Trying
To: <sip:48485858 at 10.3.8.20:5060>
From: "Unknown" <sip:38531300 at dialer2.krs>;tag=65480
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK117c679a7a564bb8609b56e88,
SIP/2.0/TCP
10.3.8.19:5062;branch=z9hG4bK54b1106da2bc4a3d07f6ace9a;rport=4283;received=80.239.13.209
Call-ID: 271c333045c13cf007cb1519b5c09c6f at 10.3.8.19
CSeq: 1 INVITE
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0
SIP/2.0 183 Session Progress
To: <sip:48485858 at 10.3.8.20:5060>;tag=B2B.297.356
From: "Unknown" <sip:38531300 at dialer2.krs>;tag=65480
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK117c679a7a564bb8609b56e88,
SIP/2.0/TCP
10.3.8.19:5062;branch=z9hG4bK54b1106da2bc4a3d07f6ace9a;rport=4283;received=80.239.13.209
Call-ID: 271c333045c13cf007cb1519b5c09c6f at 10.3.8.19
CSeq: 1 INVITE
Record-Route: <sip:10.3.8.20:5060;lr;transport=tcp>
Content-Type: application/sdp
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Contact: <sip:85.19.212.36:5060;transport=tcp>
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 264
v=0
o=root 2116035828 2116035828 IN IP4 85.19.212.13
s=Asterisk PBX 1.6.1.20
c=IN IP4 85.19.212.13
t=0 0
m=audio 16330 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
To: <sip:48485858 at 10.3.8.20:5060>;tag=B2B.297.356
From: "Unknown" <sip:38531300 at dialer2.krs>;tag=65480
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK117c679a7a564bb8609b56e88,
SIP/2.0/TCP
10.3.8.19:5062;branch=z9hG4bK54b1106da2bc4a3d07f6ace9a;rport=4283;received=80.239.13.209
Call-ID: 271c333045c13cf007cb1519b5c09c6f at 10.3.8.19
CSeq: 1 INVITE
Record-Route: <sip:10.3.8.20:5060;lr;transport=tcp>
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Contact: <sip:85.19.212.36:5060;transport=tcp>
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0
SIP/2.0 200 OK
To: <sip:48485858 at 10.3.8.20:5060>;tag=B2B.297.356
From: "Unknown" <sip:38531300 at dialer2.krs>;tag=65480
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK117c679a7a564bb8609b56e88,
SIP/2.0/TCP
10.3.8.19:5062;branch=z9hG4bK54b1106da2bc4a3d07f6ace9a;rport=4283;received=80.239.13.209
Call-ID: 271c333045c13cf007cb1519b5c09c6f at 10.3.8.19
CSeq: 1 INVITE
Record-Route: <sip:10.3.8.20:5060;lr;transport=tcp>
Content-Type: application/sdp
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Contact: <sip:85.19.212.36:5060;transport=tcp>
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 264
v=0
o=root 2116035828 2116035829 IN IP4 85.19.212.13
s=Asterisk PBX 1.6.1.20
c=IN IP4 85.19.212.13
t=0 0
m=audio 16330 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
SIP/2.0 200 OK
To: <sip:48485858 at 10.3.8.20:5060>;tag=B2B.297.356
From: "Unknown" <sip:38531300 at dialer2.krs>;tag=65480
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK117c679a7a564bb8609b56e88,
SIP/2.0/TCP
10.3.8.19:5062;branch=z9hG4bK54b1106da2bc4a3d07f6ace9a;rport=4283;received=80.239.13.209
Call-ID: 271c333045c13cf007cb1519b5c09c6f at 10.3.8.19
CSeq: 1 INVITE
Record-Route: <sip:10.3.8.20:5060;lr;transport=tcp>
Content-Type: application/sdp
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Contact: <sip:85.19.212.36:5060;transport=tcp>
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 264
v=0
o=root 2116035828 2116035829 IN IP4 85.19.212.13
s=Asterisk PBX 1.6.1.20
c=IN IP4 85.19.212.13
t=0 0
m=audio 16330 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------------------------------------------------------------------
Comment By: Bogdan-Andrei Iancu (bogdan_iancu)
Date: 2011-09-29 15:33
Message:
Hi Morten,
Just tried a small test :
U 127.0.0.1:36445 -> 127.0.0.1:5060
MESSAGE sip:600 at 127.0.0.6;kk=q SIP/2.0
Via: SIP/2.0/TCP 10.3.8.20:5060;branch=z9hG4bK0440914511a92062ea12e98d4,
SIP/2.0/TCP 10.3.8.19:5062;branch=z9hG4bKaad385d13b702b61ee38b3118
...
#
U 127.0.0.1:5060 -> 127.0.0.6:5060
MESSAGE sip:600 at 127.0.0.6;kk=q SIP/2.0
Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK7218.b5582ab096480158f0068e6fe2b7eba3.0
Via: SIP/2.0/TCP
10.3.8.20:5060;rport=36445;received=127.0.0.1;branch=z9hG4bK0440914511a92062ea12e98d4,
SIP/2.0/TCP 10.3.8.19:5062;branch=z9hG4bKaad385d13b702b61ee38b3118
And as you can see, the rport and received were inserted to the right body
part.
Could you upload a full message trace (in and out) showing the problem you
reported ?
Regards,
Bogdan
----------------------------------------------------------------------
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