asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	30cd559345	DNS: Need to use the same serializer for a pjproject SIP transaction. All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject when using an external DNS resolver to process messages for the transaction. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_resolver.c use the requesting thread's serializer to execute the async callback. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. ASTERISK-25115 #close Reported by: John Bigelow Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a	11 years ago
Kevin Harwell	e62bd46511	res_pjsip: make it unloadable (take 2) Due to the original patch causing memory corruptions it was removed until the problem could be resolved. This patch is the original patch plus some added locking around stasis router subcription that was needed to avoid the memory corruption. Description of the original problem and patch (still applicable): The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4363/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) ........ Merged revisions 431179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	07e2a48ab1	REVERTING res_pjsip: make it unloadable Due to the original patch causing memory corruptions the patch is being removed until the problem can be resolved. ........ Merged revisions 430734 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	49542a794b	res_pjsip: make it unloadable The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4311/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) ........ Merged revisions 430628 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	47074f4bfd	res_pjsip: Ensure in-dialog responses have an endpoint associated. When handling incoming messages we determine if it is associated with a dialog. If so we use that to determine what serializer and endpoint to use for the message. Previously this would pass the endpoint to the endpoint lookup module to actually place the endpoint completely on the message. For in-dialog responses, however, this did not occur as dialog processing took over and the endpoint lookup did not occur. This change just places the endpoint in the expected spot immediately instead of relying on the endpoint lookup module. In-dialog responses thus have the expected endpoint. AST-1459 #close Review: https://reviewboard.asterisk.org/r/4146/ ........ Merged revisions 427490 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427491 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	45b7b474ac	res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible (although it shouldn't occur) for pjproject to pass up an rdata object with a NULL msg in the msg_info. Needless to say, things that attempt to dereference this are in for a rough ride. In particular, this caused crashes in three different locations, all of which are 'low level' enough to intercept an rdata object early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) res_pjsip/distributor Anything that can intercept an rdata object before res_pjsip/distributor should be defensive when looking at the received packet. #SIPit31 ASTERISK-24369 #close Reported by: Matt Jordan ........ Merged revisions 424618 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424619 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7cbb6eab15	PJSIP: Add Path header support This adds Path support to chan_pjsip in res_pjsip_path.c with minimal additions in res_pjsip_registrar.c to store the path and additions in res_pjsip_outbound_registration.c to enable advertisement of path support to registrars and intervening proxies. Path information is stored on contacts and is enabled via Address of Record (AoRs) and Registration configuration sections. While adding path support, it became necessary to be able to add SIP supplements that handled messages outside of sessions, so a framework for handling these types of hooks was added in parallel to the already-existing session supplements and several senders of out-of-dialog requests were refactored as a result. (closes issue ASTERISK-21084) Review: https://reviewboard.asterisk.org/r/3050/ ........ Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	aced8bdd2e	Fix erroneous behavior when sending auth rejection to artificial endpoint. We were not including an authentication challenge when sending a 401 response to unmatched endpoints. This was due to the conversion to use a vector for authentication section names on an endpoint. The vector for artificial endpoints was empty, resulting in the challenge being sent back containing no challenges. This is worked around by placing a bogus value in the artificial endpoint's auth vector. This value is never looked up by anything, since they instead will directly call ast_sip_get_artificial_auth(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	b18ed67d16	Switch PJSIP auth to use a vector. Since Asterisk has a vector API now, places where arrays are manually resized don't really make sense any more. Since the auth work in PJSIP was freshly-written, it was easy to reform it to use a vector. Review: https://reviewboard.asterisk.org/r/3044 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	1b14a78d14	res_pjsip: Add support for building against pjproject with SIP transaction group lock support. SIP transaction group lock support has been backported into our pjproject. Since the code now internally uses a group lock the code is now changed to unlock it if present. Note that the act of finding the transaction is what actually returns it locked. For further information about group locks check out the wiki page at: http://trac.pjsip.org/repos/wiki/Group_Lock (issue ASTERISK-22818) Reported by: Matt Jordan ........ Merged revisions 402864 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	bb1568caa1	PJSIP: Add log messages when requests are received for non-existent endpoints (closes issue ASTERISK-22552) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2934/ ........ Merged revisions 401938 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	a181544299	Fix dialog matching in the SIP distributor. Dialog matching is performed in the distributor for the sole purpose of retrieving an associated serializer so the request may be serialized. This patch fixes two problems. First, incoming CANCEL requests that had no to-tag (which really should be all CANCEL requests) would not match with a dialog. An earlier bug fix to deal with early CANCEL requests would result in the CANCEL being replied to with a 481. The fix for this is to find the matching INVITE transaction and get the dialog from that transaction. Second, no SIP responses were matching dialogs. This is because we were inverting the tags that we were passing into PJSIP's dialog finding function. This logic has been corrected by setting local and remote tag variables based on whether the incoming message is a request or response. ........ Merged revisions 397854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_" are now "chan_pjsip_" * All files that were "res_sip" are now "res_pjsip" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_" are now "pjsip_" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

13 Commits (40caf0ad9bef07bdfb568a88192c157dd1840100)