[asterisk-bugs] [Asterisk 0004903]: [patch] SIP over TCP project
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Sep 26 06:06:27 CDT 2007
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=4903
======================================================================
Reported By: hjlee
Assigned To: oej
======================================================================
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:
======================================================================
Date Submitted: 08-05-2005 00:41 CDT
Last Modified: 09-26-2007 06:06 CDT
======================================================================
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
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0004904 [patch] SIP over TCP project
related to 0010354 Add Basic Support For RFC 4662 (Subscri...
======================================================================
----------------------------------------------------------------------
mvanbaak - 09-26-07 06:06
----------------------------------------------------------------------
alphaque: Can you provide a patch against current svn trunk ?
As serge-v stated several times this is a new feature and should be
developed for trunk, not a release (1.2 or 1.4)
Issue History
Date Modified Username Field Change
======================================================================
09-26-07 06:06 mvanbaak Note Added: 0071102
======================================================================
More information about the asterisk-bugs
mailing list