You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.

Joshua Colp 25cf186b5f Multiple revisions 402345,405234,409129-409130,409565,413008,417141,417677 ... ........ r402345 | kmoore | 2013-11-01 05:31:49 -0700 (Fri, 01 Nov 2013) | 11 lines chan_sip: Fix RTCP port for SRFLX ICE candidates This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ ........ r405234 | kharwell | 2014-01-09 08:49:55 -0800 (Thu, 09 Jan 2014) | 19 lines res_rtp_asterisk: Fails to resume WebRTC call from hold In ast_rtp_ice_start if the ice session create check list failed, start check was never initiated and ice_started was never set to true. Upon re-entering the function (for instance, [un]hold) it would try to create the check list again with duplicate remote candidates. Fixed so that if the create check list fails the necessary data structures are properly re-initialized for any subsequent retries. Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a check list failure because it possible things might still work. However, a debug message was added to help with any future troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Patches: works_on_my_machine.patch uploaded by xytis (license 6558) ........ r409129 | jrose | 2014-02-27 11:19:02 -0800 (Thu, 27 Feb 2014) | 15 lines res_rtp_asterisk: Fix checklist creating problems in ICE sessions Prior to this patch, local candidate lists including SRFLX would fail to start properly when building ICE candidate check lists. This patch fixes that problem by making sure that each SRFLX candidate is associated with the proper base address so that the check list can create matches properly. This patch was written by jcolp. The issue will be left open to await testing by the issue participants. (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose | 2014-02-27 11:38:10 -0800 (Thu, 27 Feb 2014) | 8 lines res_rtp_asterisk: correct build error from r409129 Accidentally placed a declaration below functional code (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ ........ r409565 | jrose | 2014-03-04 08:40:39 -0800 (Tue, 04 Mar 2014) | 9 lines res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE ICE sessions will now be restarted if sessions are changed to use new sets of remote candidates. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/ ........ r413008 | mjordan | 2014-04-25 10:47:21 -0700 (Fri, 25 Apr 2014) | 14 lines res_rtp_asterisk: Add support for DTLS handshake retransmissions On congested networks, it is possible for the DTLS handshake messages to get lost. This patch adds a timer to res_rtp_asterisk that will periodically check to see if the handshake has succeeded. If not, it will retransmit the DTLS handshake. Review: https://reviewboard.asterisk.org/r/3337 ASTERISK-23649 #close Reported by: Nitesh Bansal patches: dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418) ........ r417141 | file | 2014-06-23 11:49:14 -0700 (Mon, 23 Jun 2014) | 5 lines res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0. ASTERISK-23834 #close Reported by: Richard Kenner ........ r417677 | file | 2014-06-30 12:42:18 -0700 (Mon, 30 Jun 2014) | 12 lines 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 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/ ........ Merged revisions 402345,405234,409129-409130,409565,413008,417141,417677 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/11.6@417724 65c4cc65-6c06-0410-ace0-fbb531ad65f3		12 years ago
..
include	Multiple revisions 402345,405234,409129-409130,409565,413008,417141,417677	12 years ago
config_parser.c	Multiple revisions 370769-370771	14 years ago
dialplan_functions.c	Fix refcounting of sip_pvt in test_sip_rtpqos test and unlink it from the list of pvts.	13 years ago
reqresp_parser.c	Fix Record-Route parsing for large headers.	13 years ago
sdp_crypto.c	Corrected crypto tag in SDP ANSWER for SRTP. (again)	13 years ago
security_events.c	AST-2013-003: Prevent username disclosure in SIP channel driver	13 years ago
srtp.c	Multiple revisions 369001-369002	14 years ago
utils.c	Multiple revisions 369001-369002	14 years ago