1. Brekeke Product Name and version: Brekeke SIP Server , Version 2.4.8.6 Evaluation
2. Java version: 5
3. OS type and the version: Windows XP
4. UA (phone), gateway or other hardware/software involved:
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : 1
6. Your problem:
I have the following problem when using Brekeke SIP Server over TCP.
My implementation receive an INVITE from Brekeke:
INVITE sip:joe@10.130.4.160:5060 SIP/2.0
Via: SIP/2.0/TCP 10.130.4.138:4846;rport;branch=z9hG4bK7a41623fa6535a8966be-7f3e03e-2baf5f79
Via: SIP/2.0/TCP 10.130.1.29:5060;branch=z9hG4bK04EADA2D3B1B3C7678DF9F1D185597BC8;rport=1443
From: "Your Full Name" <sip:Username@defaultproxy:5060>;tag=7C32508DD83F649D2CA056C061F3CE6B
To: <sip:joe@10.130.4.138>
Contact: <sip:Username@10.130.4.138:4846;transport=tcp>
Call-ID: 273D07F280DC2195022081C3EB4364EB1B29C@10.130.4.138:5060
User-Agent: Kapanga Softphone Desktop Windows 1.00/2182d+1229993678_000F1FC88160 (not registered)
Supported: timer, replaces
CSeq: 1 INVITE
Max-Forwards: 69
Session-Expires: 1800;refresher=uac
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Record-Route: <sip:10.130.4.138:4846;transport=tcp;lr>
Content-Type: application/sdp
Content-Length: 384
...
from which the Contact and Record-Route headers mention to use port 4846. But then this port is unreachable (I checked with netstat, and I did not see a listen on that port). I think that this is an error and that Brekeke should specifies 5060. If really 4846 should be used, then Brekeke should listen on that port, but this is not the case. This makes my implementation to fail to send a BYE.
This looks similar to something that has been fixed in release 2.4.8.6 "- Fixed the bug where the registrar may not point reachable port number if a user registers over TCP. "
Anybody knows if this is a bug and if it will be fixed ?
Thanks,
Pierre
Unreachable port when using TCP.
Moderator: Brekeke Support Team
set &net.sip.transport.follow.request = true in the DialPlan's Deploy Patterns.
---------------------------------------------------------------------
http://wiki.brekeke.com/wiki/TCP-Support-Details
net.sip.transport.follow.request: default: false; if true, a SIP request packet will be sent over the same transport of the initial request packet.
---------------------------------------------------------------------
---------------------------------------------------------------------
http://wiki.brekeke.com/wiki/TCP-Support-Details
net.sip.transport.follow.request: default: false; if true, a SIP request packet will be sent over the same transport of the initial request packet.
---------------------------------------------------------------------
Thank you Haddas,
But this does not work. I tried also &net.sip.tcp.reuse=false without success.
I think that both these patterns are for requests from Brekeke.
My issue is when the initial request (INVITE) comes from Brekeke. And then my application wants to sent a new request (BYE) in the opposite direction. To do so, it needs to use the address from the Contact header, or from the Record-Route header if present:
Brekeke <----> My application
----> INVITE
<---- 200 OK
----> ACK
<---- BYE (cannot open TCP connection using the Contact or Record-Route headers' port.
INVITE sip:joe@10.130.4.160:5060 SIP/2.0
Via: SIP/2.0/TCP 10.130.4.138:4846;rport;branch=z9hG4bK7a41623fa6535a8966be-7f3e03e-2baf5f79
Via: SIP/2.0/TCP 10.130.1.29:5060;branch=z9hG4bK04EADA2D3B1B3C7678DF9F1D185597BC8;rport=1443
From: "Your Full Name" <sip:Username@defaultproxy:5060>;tag=7C32508DD83F649D2CA056C061F3CE6B
To: <sip:joe@10.130.4.138>
Contact: <sip:Username@10.130.4.138:4846;transport=tcp>
Call-ID: 273D07F280DC2195022081C3EB4364EB1B29C@10.130.4.138:5060
User-Agent: Kapanga Softphone Desktop Windows 1.00/2182d+1229993678_000F1FC88160 (not registered)
Supported: timer, replaces
CSeq: 1 INVITE
Max-Forwards: 69
Session-Expires: 1800;refresher=uac
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Record-Route: <sip:10.130.4.138:4846;transport=tcp;lr>
Content-Type: application/sdp
Content-Length: 384
...
Any other suggestion ?
Thanks,
Pierre
But this does not work. I tried also &net.sip.tcp.reuse=false without success.
I think that both these patterns are for requests from Brekeke.
My issue is when the initial request (INVITE) comes from Brekeke. And then my application wants to sent a new request (BYE) in the opposite direction. To do so, it needs to use the address from the Contact header, or from the Record-Route header if present:
Brekeke <----> My application
----> INVITE
<---- 200 OK
----> ACK
<---- BYE (cannot open TCP connection using the Contact or Record-Route headers' port.
INVITE sip:joe@10.130.4.160:5060 SIP/2.0
Via: SIP/2.0/TCP 10.130.4.138:4846;rport;branch=z9hG4bK7a41623fa6535a8966be-7f3e03e-2baf5f79
Via: SIP/2.0/TCP 10.130.1.29:5060;branch=z9hG4bK04EADA2D3B1B3C7678DF9F1D185597BC8;rport=1443
From: "Your Full Name" <sip:Username@defaultproxy:5060>;tag=7C32508DD83F649D2CA056C061F3CE6B
To: <sip:joe@10.130.4.138>
Contact: <sip:Username@10.130.4.138:4846;transport=tcp>
Call-ID: 273D07F280DC2195022081C3EB4364EB1B29C@10.130.4.138:5060
User-Agent: Kapanga Softphone Desktop Windows 1.00/2182d+1229993678_000F1FC88160 (not registered)
Supported: timer, replaces
CSeq: 1 INVITE
Max-Forwards: 69
Session-Expires: 1800;refresher=uac
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Record-Route: <sip:10.130.4.138:4846;transport=tcp;lr>
Content-Type: application/sdp
Content-Length: 384
...
Any other suggestion ?
Thanks,
Pierre