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Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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branch.
This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Tested by: jtodd, rjain, loloski
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This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
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Asterisk), and obfuscate email addresses enough to stop most spam harvesters.
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jabber/jingle integration
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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when the number of channels fill the MTU on a given link.
In the future, this needs to be configurable per peer with trunking enabled.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40692 | tilghman | 2006-08-20 17:09:57 -0500 (Sun, 20 Aug 2006) | 2 lines
Reformat to match the contribution style of other contributors
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credits fixed.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r39379 | kpfleming | 2006-08-08 13:39:16 -0500 (Tue, 08 Aug 2006) | 2 lines
add explicit listing of anthm's contributions (issue #7683)
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- change references to the "scx" jitterbuffer to be called "fixed" and change
references to the "stevek" jitterbuffer to be called "adaptive", instead
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patch 0006763 with minor changes.
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- with contributions from miconda, jcollie, and sb
- branch maintained by oej
Thanks everyone!
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no longer considered experimental :-)
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crash (imported from 1.2)
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