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Author	SHA1	Message	Date
Jaco Kroon	9680790531	chan_sip: improved ip:port finding of peers for non-UDP transports. ... Also remove function peer_ipcmp_cb since it's not used (according to rmudgett). Prior to b2c4e8660a (ASTERISK_27457) insecure=port was the defacto standard. That commit also prevented insecure=port from being applied for sip/tcp or sip/tls. Into consideration there are three sets of behaviour: 1. "previous" - before the above commit. 2. "current" - post above commit, pre this one. 3. "new" - post this commit. The problem that the above commit tried to address was guests over TCP. It succeeded in doing that but broke transport!=udp with host!=dynamic. This commit attempts to restore sane behaviour with respect to transport!=udp for host!=dynamic whilst still retaining the guest users over tcp. It should be noted that when looking for a peer, two passes are made, the first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer, thus looking for full matches (IP + Port), the second pass sets SIP_INSECURE_PORT, thus expecting matches on IP only where the matched peer allows for that (in the author's opinion: UDP with insecure=port, or any TCP based, non-dynamic host). In previous behaviour there was special handling for transport=tcp|tls whereby a peer would match during the first pass if the utilized transport was TCP|TLS (and the peer allowed that specific transport). This behaviour was wrong, or dubious at best. Consider two dynamic tcp peers, both registering from the same IP (NAT), in this case either peer could match for connections from an IP. It's also this behaviour that prevented SIP guests over tcp. The above referenced commit removed this behaviour, but kept applying the SIP_INSECURE_PORT only to WS|WSS|UDP. Since WS and WSS is also TCP based, the logic here should fall into the TCP category. This patch updates things such that the previously non-explicit (TCP behaviour) transport test gets performed explicitly (ie, matched peer must allow for the used transport), as well as the indeterministic source-port nature of the TCP protocol is taken into account. The new match algorithm now looks like: 1. As per previous behaviour, IP address is matched first. 2. Explicit filter with respect to transport protocol, previous behaviour was semi-implied in the test for TCP pure IP match - this now made explicit. 3. During first pass (without SIP_INSECURE_PORT), always match on port. 4. If doing UDP, match if matched against peer also has SIP_INSECURE_PORT, else don't match. 5. Match if not a dynamic host (for non-UDP protocols) 6. Don't match if this is WS|WSS, or we can't trust the Contact address (presumably due to NAT) 7. Match (we have a valid Contact thus if the IP matches we have no choice, this will likely only apply to non-NAT). To logic-test this we need a few different scenarios. Towards this end, I work with a set number of peers defined in sip.conf: [peer1] host=1.1.1.1 transport=tcp [peer2] host=1.1.1.1 transport=udp [peer3] host=1.1.1.1 port=5061 insecure=port transport=udp [peer4] host=1.1.1.2 transport=udp,tcp [peer5] host=dynamic transport=udp,tcp Test cases for UDP: 1 - incoming UDP request from 1.1.1.1: - previous: - pass 1: * peer1 or peer2 if from port 5060 (indeterminate, depends on peer ordering) * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 - current: as per previous. - new: - pass 1: * peer2 if from port 5060 * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 2 - incoming UDP request from 1.1.1.2: - previous: - pass 1: * peer5 if registered from 1.1.1.2 and port matches * peer4 if source port is 5060 - pass 2: * no match (guest) - current: as previous. - new as previous (with the variation that if peer5 didn't have udp as allowed transport it would not match peer5 whereas previous and current code could). 3 - incoming UDP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address and source port matches. - pass 2: * peer5 if insecure=port is additionally set. * no match (guest) - current - as per previous - new - as per previous Test cases for TCP based transports: 4 - incoming TCP request from 1.1.1.1 - previous: - pass 1 (indeterministic, depends on ordering of peers in memory): * peer1; or * peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or * peer2 if the source port happens to be 5060; or * peer3 if the source port happens to be 5061. - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer1 or peer2 if from source port 5060 * peer3 if from source port 5060 * peer5 if registered as 1.1.1.1 and source port matches - pass 2: * no match (guest) - new: - pass 1: * peer 1 if from port 5060 * peer 5 if registered and source port matches - pass 2: * peer 1 5 - incoming TCP request from 1.1.1.2 - previous (indeterminate, depends on ordering): - pass 1: * peer4; or * peer5 if peer5 registered from 1.1.1.2 - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * no match (guest). - new: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * peer4 6 - incoming TCP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer5 if registered from that address and port matches. - pass 2: * no match (guest) - new: as per current. It should be noted the test cases don't make explicit mention of TLS, WS or WSS. WS and WSS previously followed UDP semantics, they will now enforce source port matching. TLS follow TCP semantics. The previous commit specifically tried to address test-case 6, but broke test-cases 4 and 5 in the process. ASTERISK-27881 #close Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2	7 years ago
Jaco Kroon	a74f8e51a6	AMI: be less verbose when adding HTTP headers to AMI/HTTP messages. ... All HTTP/AMI message headers are being sent to the verbose channel. There are multiple places this is happening. Consolidate the loop into a function. Drop the debug/verbose message. Convert to using ast_asprintf to perform the length calculation, memory allocation and snprintf all in one step. Change-Id: Ic45e673fde05bd544be95ad5cdbc69518207c1a1	7 years ago
Jenkins2	8b9f01342b	Merge "sample_configs: noload res_hep.so by default"	7 years ago
Florian Floimair	3bdbbb7637	alembic: increase uri column size ... When mobile SIP clients register with Asterisk that use some sort of push notifications, the URI can get quite lengthy due to the additional push-service annotations (things like tokens, pn-type, etc.) contained in it. ASTERISK-28022 #close Change-Id: I4c7ceadc3bb405f3daf722641c8cd5ca4188cc37	7 years ago
Matthew Fredrickson	c8bacd45f1	sample_configs: noload res_hep.so by default ... Change disables loading of res_hep.so in default installation. Loading res_hep has a performance impact whether it's used or not. This disables loading of it in sample config files. Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0	7 years ago
Joshua Colp	5320b18bfe	Merge "res_pjsip: Reduce processing when a Contact is updated."	7 years ago
Sean Bright	14c6f8be9d	app_queue: Silence GCC 8 compiler warning ... I'm only seeing an error in 14+, so I assume it is due to different compiler options: app_queue.c: In function ‘handle_queue_add_member’: app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11 bytes into a region of size 3 [-Werror=format-overflow=] sprintf(num, "%d", state); ^~ app_queue.c:10234:18: note: directive argument in the range [-2147483648, 99] sprintf(num, "%d", state); ^~~~ Compiler: gcc version 8.0.1 20180414 (experimental) [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2) Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10	7 years ago
Joshua Colp	a4d1c4dabe	Merge "AMI: Remove docs for nonexistent AMI ContactStatus event headers"	7 years ago
Joshua Colp	2dce2c9796	Merge "pbx_dundi: Fix debug frame decode string."	7 years ago
George Joseph	55f72f21b3	Merge "pbx_dundi.c: Handle thread shutdown better."	7 years ago
Joshua Colp	61304d7b26	Merge "pbx_dundi.c: Misc memory management fixes when destroying peers"	7 years ago
Richard Mudgett	5ec27d5206	AMI: Remove docs for nonexistent AMI ContactStatus event headers ... Change-Id: I5736965c64c44338f7330e85a24bb46818607f19	7 years ago
George Joseph	96363e542b	Merge "res_rtp_asterisk.c: Fix unused variable warnings"	7 years ago
George Joseph	27d94dc70d	Merge "res_sorcery_realtime.c: Fix unqualified fetch warning."	7 years ago
George Joseph	d22d761257	Merge "res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response."	7 years ago
Joshua Colp	457ba355aa	res_pjsip: Reduce processing when a Contact is updated. ... When a Contact is updated the only material change that qualify support cares about is the underlying configuration for the AOR. In this case we will update things with the new AOR information but otherwise the callback to indicate the Contact has changed can be ignored. This is because it is only when a Contact is added or deleted that material changes occur within the qualify support. An update can't change the URI since it would result in a new Contact so it can be ignored. Change-Id: I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d	7 years ago
Richard Mudgett	40f1604e2f	res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response. ... We were still getting crashes after the first fix. Somehow we receive a non-2xx final response before we get a 200 final response. With the failure response we had already cleaned up and destroyed some data structures. When the unexpected 200 response comes in we crash. * Add protection code to prevent processing another final T.38 reINVITE response. ASTERISK-27944 Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74	7 years ago
Richard Mudgett	8cd36ab9b6	res_sorcery_realtime.c: Fix unqualified fetch warning. ... The allow_unqualified_fetch option for the sorcery realtime backend blocked actually fetching all rows when the option is set to warn. * Made issue a warning and actually do the request when allow_unqualified_fetch=warn is set. Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312	7 years ago
Kirsty Tyerman	328f772d3b	pbx_dundi: Added IPv6 support for dundi ... Change includes move to netsock2 library. ASTERISK-27164 Reported-by: Adam Secombe Change-Id: Ia9e8dc3d153de7a291dbda4bd87fc827dd2bb846	7 years ago
Richard Mudgett	273e2802aa	pbx_dundi.c: Misc memory management fixes when destroying peers ... * In destroy_peer(), fixed memory leaks of lookup history strings and qualify transactions when destroying peers. * In destroy_peer(), fixed leaving the registerexpire scheduled callback active when a peer is destroyed on a reload. The reload marks and sweeps peers so any peers not explicitly configured get destroyed. Peers created dynamically from the '*' peer will not exist until they re-register after the reload. These destroyed peers caused memory corruption when the registerexpire timer expired. * Made build_peer() not schedule any callbacks on the '*' peer (empty_eid). It is a special peer that is cloned to dynamically created peers so it doesn't actually get involved in any message transactions. * Made do_register_expire() remove the dundi/dpeers AstDB entry when a peer registration expires. * Fix deep_copy_peer() to not copy some things that cannot be copied to the cloned peer structure. Timers, message transactions, and lookup history are specific to a peer instance. * Made set_config() lock around processing the mappings configuration. * Reordered unload_module() to handle load_module() declining the load due to error. Change-Id: Ib846b2b60d027f3a2c2b3b563d9a83a357dce1d6	7 years ago
Richard Mudgett	d4e72ee296	pbx_dundi.c: Handle thread shutdown better. ... Change-Id: Id52f99bd6a948fe6dd82acc0a28b2447a224fe87	7 years ago
Richard Mudgett	916abe7cdc	pbx_dundi: Fix debug frame decode string. ... * Fixed a typo in the name of the REGREQ frame decode string array. * Fixed off by one range check indexing into the frame decode string array. * Removed some unneeded casts associated with the decode string array. Change-Id: I77435e81cd284bab6209d545919bf236ad7933c2	7 years ago
Richard Mudgett	c035d0afe0	pbx_dundi: Update sample config documentation. ... Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849	7 years ago
Richard Mudgett	aee5f7c1b6	res_rtp_asterisk.c: Fix unused variable warnings ... Compiling without SRTP support installed resulted in some unused variable warnings. These warnings also showed that the srtp variable was obtained and passed around some functions but not really used even when a system has SRTP installed. Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc	7 years ago
Joshua Colp	5cd416f354	Merge "res_resolver_unbound: Fix leak of config nameserver strings."	7 years ago
Joshua Colp	a88cec6334	Merge "res_pjsip: Resolve transport management leak at shutdown."	7 years ago
Kevin Harwell	b400d50b1e	Merge "res_odbc: Allow unload at shutdown."	7 years ago
George Joseph	00563ce21a	CI: Fixup for non-13 branches ... Change-Id: I5e1d4a09e58b92b541bc8ed6f9e10e54c4e5101f	7 years ago
George Joseph	e5f30eba79	CI: Final version of setting correct gerrit creds ... Change-Id: I7729ecceedceb12f52bf18dae259846aa1d993b3	7 years ago
George Joseph	8e1c541acf	CI: Add https credentials to gerrit checkouts ... If the review to be tested is in a project with restricted access, we need to use the jenkins user's gerrit https credentials when we do the checkout or the checkout will fail. Change-Id: I9dc9994763c5ebfeb9f1cff60fb53f6902b7fd5f	7 years ago
George Joseph	c2f81cf446	Merge "res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered"	7 years ago
Rodrigo Ramírez Norambuena	01c90fefb3	make config: os-release output error. ... Fix not show the error "/bin/sh: /etc/os-release: No such file or directory" when the command 'make config' is run in a System without systemv. The instruction 'make config' pre execute the syntax "$(shell . /etc/os-release && echo $$ID)" to identified if system is a Slackware and Opensuse. This change prevent show the message and is send to the /dev/null Change-Id: I7f43e281a8d9405b2519fc653de82d9b8b645fdf	7 years ago
Torrey Searle	926d647def	res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered ... If in the initial sdp the caller doesn't include the line a=rtcp-mux Then asterisk shoud not include rtcp-mux in the response regardless of rtcp-mux being enabled on the endpoint ASTERISK-28007 #close Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7	7 years ago
Corey Farrell	a83c464d9d	res_resolver_unbound: Fix leak of config nameserver strings. ... Change-Id: I3f396316bb40d1ae6e91f5f688042420f1a540ed	7 years ago
Corey Farrell	24302bda21	res_pjsip: Resolve transport management leak at shutdown. ... Cleanup idle check scheduled events at shutdown. Change-Id: I61bfbb56bac69fe840c3242927d31ff3593be461	7 years ago
Corey Farrell	eb34b881a4	res_odbc: Allow unload at shutdown. ... This makes it possible for REF_DEBUG to report no leaks when loading res_odbc. Change-Id: I1a3dea786bd6e7f4820a6dd5cbaa197fa783ce93	7 years ago
Corey Farrell	52fe5fe2c8	res_pjsip: Fix leak in pjsip_options. ... sip_options_get_endpoint_state_compositor_state leaked a reference to the first available endpoint state compositor that was found. Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7	7 years ago
George Joseph	61b6d9efa4	Merge "res_pjsip_caller_id: Add "party" parameter to RPID header."	7 years ago
Jenkins2	9241b11216	Merge "res_pjsip/rtp: No joint capabilities between streams."	7 years ago
George Joseph	76b73549db	Merge "contrib/scripts: Make astgenkey executable"	7 years ago
Jenkins2	5241a53acd	Merge "Build System: Improve ccache matching for different menuselect options."	7 years ago
Richard Mudgett	58c3677581	contrib/scripts: Make astgenkey executable ... Change-Id: I11641d65592536dea9cbca5aa94a24c25d24dd5f	7 years ago
Joshua Colp	fca3d4fe5f	res_pjsip_caller_id: Add "party" parameter to RPID header. ... This change adds the "party" parameter to the Remote-Party-ID header which indicates which party the header information is applicable to. In Asterisk this is determined on whether we are the calling or called party. This is added to improve interoperability with some implementations. ASTERISK-28006 Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca	7 years ago
Jenkins2	863858ed5a	Merge "CI: Add support for coverage processing."	7 years ago
Ben Ford	c31a01bd75	res_pjsip/rtp: No joint capabilities between streams. ... When a conference contained a mixture of audio/video and audio-only users, a NOTICE message would pop up stating there are no joint capabilities between streams. This happens because streams can never be removed, but they can be in a REMOVED state. If we have the scenario where user A joins with audio/video, user B joins with audio-only, and user C joins with audio/video, then user A leaves, the message would be triggered. That removed stream is still in the SDP, but Asterisk would pass it through, causing it to be seen as a ulaw stream. A check has been added for removed streams, setting their status to REMOVED when handling negotiated SDPs. Also addressed an issue where user A joins, then user B joins but does not receive video until much later. Full frames were not being sent, causing some PLI from the browser. Because the video was flowing in one direction, the browser sets the SSRC to 1, but Asterisk was dropping the PLI because of that. Added a check to see if the SSRC is 1 or not, which sends full frames and allows video to flow between user A and user B. This should only happen when dealing with PSFB or FUR, and in the case of PSFB, only for PLI. ASTERISK-27398 Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e	7 years ago
Ivan Poddubny	2ce061091e	app_queue: set QUEUESTATUS to LEAVEEMPTY instead of CONTINUE ... When a call leaves a queue on leaveempty condition, QUEUESTATUS must be set to LEAVEEMPTY, no matter whether Queue was executed with or without the "c" (continue) option. The regression was introduced in the fix for ASTERISK_25665. The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was overwritten in case when "c" is set, regardless of what was the cause for leaving the queue. ASTERISK-27973 #close Reported-by: Valentin Safonov Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c	7 years ago
Corey Farrell	63ca367ab9	Sample configs: Fix pjsip.conf syntax error. ... It is valid for a config file to be empty or contain only comments, but not valid for a config value to be set when no uncommented context exists. This caused an error to be loged numerous times during start when loading the default pjsip.conf. Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6	7 years ago
Joshua Colp	b0ac1ecc29	Merge "res_pjsip_registrar: Improve performance on inbound handling."	7 years ago
Corey Farrell	addfc93815	CI: Add support for coverage processing. ... Enable coverage with `./tests/CI/buildAsterisk.sh --coverage`. This will cause Asterisk to be compiled with coverage support. It also initializes 'before' coverage data for all sources. Accept --tested-only to disable modules which are not run by any test. Enabling coverage also sets tested-only true by default. To build everything with coverage enabled use `--coverage --tested-only=0`. ./tests/CI/processCoverage.sh is used to process the coverage and generate HTML reports. Fix utils/check_expr2 which failed to compiled with coverage enabled. Add status output 5 times per stage of astobj2_test_perf to ensure remote CLI does not timeout when compiled with coverage. Remote CLI disconnects if no output is received for 60 seconds. When coverage is enabled it takes about 70 seconds for my laptop to run the stages of this test, so with the change a message is printed every 14 seconds. Change-Id: I890f7d5665087426ad7d3e363187691b9afc2222	7 years ago
Joshua Colp	a0426e136b	Merge "stasis: Reduce calculation of stasis message type hash."	7 years ago