[asterisk-bugs] [Asterisk 0004903]: [patch] SIP over TCP project

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=4903 
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Reported By:                hjlee
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   4903
Category:                   Channels/chan_sip
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 46875 
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             08-05-2005 00:41 CDT
Last Modified:              09-13-2007 13:29 CDT
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Summary:                    [patch] SIP over TCP project
Description: 
I added TCP support to asterisk SIP channel. I put all my changes under
#ifdef SIP_TCP_SUPPORT and left the original code. So if you search
SIP_TCP_SUPPORT, you can find my changes very easily.

My changes
-Added TCP listening socket, siptcpsock.
-Added securechannel, sockfd, transport field to struct sip_pvt.
-Added transport, tcpsockfd field to struct sip_peer.
-Added TCP read in sipsock_read().
-Added siptcp_accept() to accept an incoming TCP connection request.
-Added transport, q parameter processing in Contact header parsing.
-Changed the hard-coded "UDP" in Via header to copy sip_pvt.transport.
-Added tcp_conenct() to make a TCP connection for outgoing message.
-Added TCP transmit in __sip_xmit().
-Saved TCP connecton socket to sip_peer.tcpsockfd, copied it to
sip_pvt.sockfd when OPTIONS or INVITE is sent to the peer that is using
TCP.

I tested it mainly xlite(UDP only free version) and Jain-SIP communicator.
call signal is working well. One problem I am having is Jain-SIP
communicator doesn't receive any audio, I don't know why. If any one has
xlite-pro(TCP supported commercial version) or TCP supported SIP clients, I
am looking forward to hear the test result.

Welcome any comment.
Thanks

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Relationships       ID      Summary
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related to          0004904 [patch] SIP over TCP project
related to          0010354 Add Basic Support For RFC 4662 (Subscri...
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 student - 09-13-07 13:29  
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Hello,

I'm experimenting exactly the same issue as "thermalwetland" stated. I
read in another forum that ip phones usually look at the dns srv entry, to
see if the server supports tls or not. Maybe that's the solution. I'll
check if it works.

Regards, 
Student 

Issue History 
Date Modified   Username       Field                    Change               
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09-13-07 13:29  student        Note Added: 0070500                          
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