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 To :  yate@v...
 From :  Radu - Eosif Mihailescu <csdexter@l...>
 Subject :  SDP rewriting/ignoring problem
 Date :  Mon, 31 May 2010 00:22:25 +0300
Hello list,

I have encountered the following problem while using Yate as a SIP PBX 
in conjunction with a well known local SIP provider that uses two 
different hosts for signalling (SIP/SDP) and media (RTP): Yate seems to 
ignore the differing address and attempts to open a RTP session with the 
signalling host -- which obviously fails.

This is how the packet in question looks like if you ask yate: {
 Received 1113 bytes SIP message from 193.16.148.244:5060
------
INVITE sip:40215697951@1...:5060 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 
193.16.148.244;branch=z9hG4bK340b.3784c6f6faf69d14f72a0f9c9f6beaba.0
Via: SIP/2.0/UDP 
193.16.148.244:5061;branch=z9hG4bK6012d095c862be98b3ec66dcbc7c20cb;rport=5061
Max-Forwards: 16
From: ;tag=6262e281464926ced47061945b8c1b68
To: 
Call-ID: 90AD75FA-6B6511DF-BA12DF22-33814A24@1...
CSeq: 200 INVITE
Contact: Anonymous 
Expires: 300
User-Agent: Sippy
cisco-GUID: 2426407105-1801785823-3109552129-1119049838
h323-conf-id: 2426407105-1801785823-3109552129-1119049838
Content-disposition: session
Content-Length: 311
Content-Type: application/sdp

v=0
o=Sippy 476310796 0 IN IP4 193.16.148.244
s=SIP Call
t=0 0
m=audio 17112 RTP/AVP 8 0 3 18 101
*c=IN IP4 193.16.148.244*
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
------
}

This is how the same packet looks like if you ask tcpdump/wireshark: {
INVITE sip:40215697951@8...:5060 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 
193.16.148.244;branch=z9hG4bK340b.3784c6f6faf69d14f72a0f9c9f6beaba.0
Via: SIP/2.0/UDP 
193.16.148.244:5061;branch=z9hG4bK6012d095c862be98b3ec66dcbc7c20cb;rport=5061
Max-Forwards: 16
From: ;tag=6262e281464926ced47061945b8c1b68
To: 
Call-ID: 90AD75FA-6B6511DF-BA12DF22-33814A24@1...
CSeq: 200 INVITE
Contact: Anonymous 
Expires: 300
User-Agent: Sippy
cisco-GUID: 2426407105-1801785823-3109552129-1119049838
h323-conf-id: 2426407105-1801785823-3109552129-1119049838
Content-disposition: session
Content-Length: 311
Content-Type: application/sdp

v=0
o=Sippy 476310796 0 IN IP4 193.16.148.244
s=SIP Call
t=0 0
m=audio 17112 RTP/AVP 8 0 3 18 101
*c=IN IP4 193.16.148.226*
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
}

I'm using Yate SVN r3342 on CentOS 5.5 i386.

If you need any additional information regarding this issue, I would be 
happy to provide it on request.

Thank you for your time,
Radu - Eosif Mihailescu

-- 
@Dexter  GSM: +40 (721) 294400
JID: csdexter@j... ICQ: 27762040 Y!: csdexter
MSN: rmihailescu@h... AIM: csdexter Skype: csdexter
Blog: http://www.linux360.ro/~csdexter/blog/


-- 
@Dexter  GSM: +40 (721) 294400
JID: csdexter@j... ICQ: 27762040 Y!: csdexter
MSN: rmihailescu@h... AIM: csdexter Skype: csdexter
Blog: http://www.linux360.ro/~csdexter/blog/




Hello list,

I have encountered the following problem while using Yate as a SIP PBX in conjunction with a well known local SIP provider that uses two different hosts for signalling (SIP/SDP) and media (RTP): Yate seems to ignore the differing address and attempts to open a RTP session with the signalling host -- which obviously fails.

This is how the packet in question looks like if you ask yate: {
<sip:INFO> Received 1113 bytes SIP message from 193.16.148.244:5060
------
INVITE sip:40215697951@1...:5060 SIP/2.0
Record-Route: <sip:193.16.148.244;ftag=6262e281464926ced47061945b8c1b68;lr>
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bK340b.3784c6f6faf69d14f72a0f9c9f6beaba.0
Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bK6012d095c862be98b3ec66dcbc7c20cb;rport=5061
Max-Forwards: 16
From: <sip:0733069042@1...>;tag=6262e281464926ced47061945b8c1b68
To: <sip:40215697951@1...>
Call-ID: 90AD75FA-6B6511DF-BA12DF22-33814A24@1...
CSeq: 200 INVITE
Contact: Anonymous <sip:193.16.148.244:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 2426407105-1801785823-3109552129-1119049838
h323-conf-id: 2426407105-1801785823-3109552129-1119049838
Content-disposition: session
Content-Length: 311
Content-Type: application/sdp

v=0
o=Sippy 476310796 0 IN IP4 193.16.148.244
s=SIP Call
t=0 0
m=audio 17112 RTP/AVP 8 0 3 18 101
c=IN IP4 193.16.148.244
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
------

}

This is how the same packet looks like if you ask tcpdump/wireshark: {
INVITE sip:40215697951@8...:5060 SIP/2.0
Record-Route: <sip:193.16.148.244;ftag=6262e281464926ced47061945b8c1b68;lr>
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bK340b.3784c6f6faf69d14f72a0f9c9f6beaba.0
Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bK6012d095c862be98b3ec66dcbc7c20cb;rport=5061
Max-Forwards: 16
From: <sip:0733069042@1...>;tag=6262e281464926ced47061945b8c1b68
To: <sip:40215697951@1...>
Call-ID: 90AD75FA-6B6511DF-BA12DF22-33814A24@1...
CSeq: 200 INVITE
Contact: Anonymous <sip:193.16.148.244:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 2426407105-1801785823-3109552129-1119049838
h323-conf-id: 2426407105-1801785823-3109552129-1119049838
Content-disposition: session
Content-Length: 311
Content-Type: application/sdp

v=0
o=Sippy 476310796 0 IN IP4 193.16.148.244
s=SIP Call
t=0 0
m=audio 17112 RTP/AVP 8 0 3 18 101
c=IN IP4 193.16.148.226
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive

}

I'm using Yate SVN r3342 on CentOS 5.5 i386.

If you need any additional information regarding this issue, I would be happy to provide it on request.

Thank you for your time,
Radu - Eosif Mihailescu
-- 
@Dexter <radu.mihailescu@l...> GSM: +40 (721) 294400
JID: csdexter@j... ICQ: 27762040 Y!: csdexter
MSN: rmihailescu@h... AIM: csdexter Skype: csdexter
Blog: http://www.linux360.ro/~csdexter/blog/

-- 
@Dexter <radu.mihailescu@l...> GSM: +40 (721) 294400
JID: csdexter@j... ICQ: 27762040 Y!: csdexter
MSN: rmihailescu@h... AIM: csdexter Skype: csdexter
Blog: http://www.linux360.ro/~csdexter/blog/