Commit Graph

6 Commits

Author SHA1 Message Date
Alexander Traud 9440f6ec58 main: Fix for Doxygen.
ASTERISK-29763

Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e
2021-12-02 15:02:09 -06:00
Alexander Traud 57fef28dc9 progdocs: Avoid 'name' with Doxygen \file.
Fixes four misuses of the parameter 'name'. Additionally, for
consistency and to avoid such an issue in future, those few other
places, which used '\file name', were changed just to '\file'. Then,
Doxygen uses the name of the current file.

ASTERISK-29733

Change-Id: I0c18b4c863c6988b138c77448057349a9ee7052d
2021-11-18 08:17:56 -06:00
Richard Mudgett 41aba83ff6 res_srtp: Move SDP SRTP code from the core to res_srtp.
A patch made to the master branch (Now the 14 branch) inadvertently made
libsrtp a required dependency in order to compile Asterisk.  Rather than
create dummy defines to substitute for the defines supplied by libsrtp
when libsrtp is not available, most of the code in sdp_srtp.c is moved
into res_srtp.c.  This gets more code out of Asterisk's core that isn't
used when SRTP is not available.  This also makes another inadvertent
required dependency on libsrtp by Asterisk's core unlikely.

ASTERISK-26253 #close
Reported by: Ben Merrills

Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7
2016-08-10 17:43:15 -05:00
Alexander Traud 1d2173c7ae res_srtp: Enable AES-256 and AES-GCM.
ASTERISK-26190 #close

Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b
2016-07-21 16:25:41 +02:00
Joshua Colp 6e60f5d317 Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11
........
res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.

This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide
a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation
completes. Configuration options to chan_sip and chan_pjsip have also been added to
allow behavior to be tweaked (such as forcing the AVP type media transports in SDP).

ASTERISK-22961 #close
Reported by: Jay Jideliov

Review: https://reviewboard.asterisk.org/r/3679/
Review: https://reviewboard.asterisk.org/r/3686/
........

Merged revisions 417678 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-30 19:51:28 +00:00
Joshua Colp 77002bc377 Merge in current pimp_my_sip work, including:
1. Security events
2. Websocket support
3. Diversion header + redirecting support
4. An anonymous endpoint identifier
5. Inbound extension state subscription support
6. PIDF notify generation
7. One touch recording support (special thanks Sean Bright!)
8. Blind and attended transfer support
9. Automatic inbound registration expiration
10. SRTP support
11. Media offer control dialplan function
12. Connected line support
13. SendText() support
14. Qualify support
15. Inband DTMF detection
16. Call and pickup groups
17. Messaging support

Thanks everyone!

Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-22 14:03:22 +00:00