17 Commits (10)

Author	SHA1	Message	Date
Jonathan Rose	6c2c8d5a58	res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session ... When srtp_create fails, the session may be dealloced or just not alloced. At the same time though, the session pointer might not be set to NULL in this process and attempting to srtp_dealloc it again will cause a segfault. This patch checks for failure of srtp_create and sets the session pointer to NULL if it fails. (closes issue ASTERISK-20499) Reported by: tootai Review: https://reviewboard.asterisk.org/r/2228/ ........ Merged revisions 377256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@377261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	9dbc175305	Fix a variety of memory leaks ... This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@366881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	455c5c8c66	Add missing newlines to CLI logging ... ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@361472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ff7665db86	Remove srtp_shutdown from res_srtp ... The patch for ASTERISK-19253 included properly shutting down the libsrtp library in the case of module unload. Unfortunately, not all distributions have the srtp_shutdown call. As such, this patch removes calling srtp_shutdown. ........ Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e796b45951	Allow SRTP policies to be reloaded ... Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	88af121f9f	Audit of ao2_iterator_init() usage for v1.8. ... Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged revisions 352955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@352956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	176819727b	Add a separate buffer for SRTCP packets ... The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) ........ Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	34ae017509	Merged revisions 337541 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	Merged revisions 328209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	085b7b212a	Merged revisions 318919 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) | 10 lines This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too much time has passed between sending audio. (closes issue #18206) Reported by: bernhardsi Patches: res_srtp_unhold.patch uploaded by bernhards (license 1138) Tested by: bernhards, notthematrix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Andrew Latham	9f1a17f137	Replacing doc/* with wiki links ... Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	9653b5d500	Merged revisions 292309 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) | 10 lines Add sip show peer info about crypto and remove dated comment This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c81da53206	Merged revisions 292016 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) | 5 lines Ref/unref res_srtp when we create/destroy a session This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp tries to unload before chan_sip does. Thanks, Russell! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	a51ce289b2	Merged revisions 287056 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) | 10 lines Don't hang up a call on an SRTP unprotect failure Also make it more obvious when there is an issue en/decrypting. (closes issue #17563) Reported by: Alexcr Patches: res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by: twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	920f5ea8b7	Merged revisions 284477 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP lines Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk ... After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago