asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	affe7ee879	progdocs: Fix grouping for latest Doxygen. Since Doxygen 1.8.16, a special comment block is required. Otherwise (pure C comment), the group command is ignored. Additionally, several unbalanced group commands were fixed. ASTERISK-29732 Change-Id: I4687857b9d56e6f44fd440b73af156691660202e	4 years ago
Naveen Albert	24a04054ad	documentation: Standardize examples Most examples in the XML documentation use the example tag to demonstrate examples, which gets parsed specially in the Wiki to make it easier to follow for users. This fixes a few modules to use the example tag instead of vanilla para tags to bring them in line with the standard syntax. ASTERISK-29777 #close Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b	4 years ago
Alexander Traud	726d6dd166	channels: Fix for Doxygen. ASTERISK-29762 Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7	4 years ago
Josh Soref	d46ba42910	channels: Spelling fixes Correct typos of the following word families: appease permanently overriding residue silliness extension channels globally reference japanese group coordinate registry information inconvenience attempts cadence payloads presence provisioning mimics behavior width natively syslabel not owning unquelch mostly constants interesting active unequipped brodmann commanding backlogged without bitstream firmware maintain exclusive practically structs appearance range retransmission indication provisional associating always whether cyrillic distinctive components reinitialized initialized capability switches occurring happened outbound ASTERISK-29714 Change-Id: Ife52ee89cd2170b684fa651ca72b1cb911a57339	4 years ago
Naveen Albert	0ca3ebe7cd	chan_alsa, chan_sip: Add replacement to moduleinfo Adds replacement modules to the moduleinfo for chan_alsa and chan_sip. ASTERISK-29601 #close Change-Id: I7a4877b0d5c0c17e088e8fa8ebbfa9a195223cbc	4 years ago
Joshua C. Colp	93870e7bb4	policy: Deprecate modules and add versions to others. app_meetme is deprecated in 19, to be removed in 21. app_osplookup is deprecated in 19, to be removed in 21. chan_alsa is deprecated in 19, to be removed in 21. chan_mgcp is deprecated in 19, to be removed in 21. chan_skinny is deprecated in 19, to be removed in 21. res_pktccops is deprecated in 19, to be removed in 21. app_macro was deprecated in 16, to be removed in 21. chan_sip was deprecated in 17, to be removed in 21. res_monitor was deprecated in 16, to be removed in 21. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29558 ASTERISK-29567 ASTERISK-29572 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Naveen Albert	fd40752954	chan_sip: Expand hook flash recognition. Some ATAs send hook flash events as application/hook-flash, rather than a DTMF event. Now, we also recognize hook-flash as a flash event. ASTERISK-29370 Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725	4 years ago
George Joseph	09303e8e22	Updates for the MessageSend Dialplan App Enhancements: * The MessageSend dialplan application now takes an optional third argument that can set the message's "To" field on outgoing messages. It's an alternative to using the MESSAGE(to) dialplan function. NOTE: No channel driver currently implements this field. A follow-on commit for res_pjsip_messaging will implement it for the chan_pjsip channel driver. * To prevent confusion with the first argument, currently named "to", it's been renamed to "destination". Its function, creating the request URI, hasn't changed. * The documentation for MessageSend was updated to be more clear about the parameters and how they interact the MESSAGE() dialplan function. * With the rename of MessageSend's first parameter, and the fact that message.c references <info> elements in chan_sip.c, res_pjsip_messaging.c and res_xmpp, they each needed documentation updates to use MessageDestinationInfo instead of MessageToInfo. * appdocsxml.dtd was updated to include a missing element declaration for "dataType". This was showing up as an error in Eclipse's dtd editor. * Despite the changes in this commit, there should be no impact to current users of MessageSend. Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a	4 years ago
Alexander Traud	5894535fed	chan_sip: Filter pass-through audio/video formats away, again. Instead of looking for pass-through formats in the list of transcodable formats (which is going to find nothing), go through the result which is going to be the jointcaps of the tech_pvt of the channel. Finally, only with that list, ast_format_cap_remove(.) is going to succeed. This restores the behaviour of Asterisk 1.8. However, it does not fix ASTERISK_29282 because that issue report is about chan_sip and PJSIP. Here, only chan_sip is fixed because PJSIP does not even call ast_rtp_instance_available_formats -> ast_translate_available_format. Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34	4 years ago
Alexander Traud	1f77c33c02	chan_sip: Allow [peer] without audio (text+video). Two previous commits, `620d9f4` and `6d980de`, allow to set up a call without audio, again. That was introduced originally with commit `f04d5fb` but changed and broke over time. The original commit missed one scenario: A [peer] section in sip.conf, which does not allow audio at all. In that case, chan_sip rejected the call, although even when the requester offered no audio. Now, chan_sip does not check whether there is no audio format but checks whether there is no format in general. In other words, if there is at least one format to offer, the call succeeds. However, to prevent calls with no-audio, chan_sip still rejects calls when both call parties (caller = requester of the call and callee = [peer] section in sip.conf) included audio. In such a case, it is expected that the call should have audio. ASTERISK-29280 Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e	4 years ago
Alexander Traud	620d9f4782	chan_sip: Set up calls without audio (text+video), again. The previous commit `6d980de` fixed this issue in the core of Asterisk. With that, each channel technology can be used without audio theoretically. Practically, the channel-technology driver chan_sip turned out to have an invalid check preventing that. chan_sip tested whether there is at least one audio format. However, chan_sip has to test whether there is at least one format. More cannot be tested while requesting chan_sip because only the [general] capabilities but not the [peer] caps are known yet. And the [peer] caps might not be a subset or show any intersection with the [general] caps. This change here fixes this. The original commit `f04d5fb`, thirteen years ago, contained a software bug as it passed ANY audio capability to the channel-technology driver. Instead, it should have passed NO audio format. Therefore, this addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8. Then, Asterisk 10 changed that from ANY to NO, but nobody reported since then. ASTERISK-29265 Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f	4 years ago
Alexander Traud	4aff42b274	chan_sip: SDP: Reject audio streams correctly. This completes the fix for ASTERISK_24543. Only when the call is an outgoing call, consult and append the configured format capabilities (p->caps). When all audio formats got rejected the negotiated format capabilities (p->jointcaps) contain no audio formats for incoming calls. This is required when there are other accepted media streams. ASTERISK-29258 Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9	4 years ago
Alexander Traud	1c05667cfc	chan_sip: SDP: Sidestep stream parsing when its media is disabled. Previously, chan_sip parsed all known media streams in an SDP offer like video (and text) even when videosupport=no (and textsupport=no). This wasted processor power. Furthermore, chan_sip accepted SDP offers, including no audio but just video (or text) streams although videosupport=no (or textsupport=no). Finally, chan_sip denied the whole offer instead of individual streams when they had encryption (SDES-sRTP) unexpectedly enabled. ASTERISK-29238 ASTERISK-29237 ASTERISK-29222 Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755	4 years ago
Alexander Traud	103d7da3bb	chan_sip: Remove unused sip_socket->port. 12 years ago, with ASTERISK_12115 the last four get/uses of socket.port vanished. However, the struct member itself and all seven set/uses remained as dead code. ASTERISK-28798 Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59	5 years ago
Alexander Traud	28faafd1c4	Compiler fixes for GCC when printf %s is NULL ASTERISK-29146 Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41	5 years ago
Alexander Traud	cd32317691	chan_sip: On authentication, pick MD5 for sure. RFC 8760 added new digest-access-authentication schemes. Testing revealed that chan_sip does not pick MD5 if several schemes are offered by the User Agent Server (UAS). This change does not implement any of the new schemes like SHA-256. This change makes sure, MD5 is picked so UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can still be used. This should have worked since day one because SIP/2.0 already envisioned several schemes (see RFC 3261 and its augmented BNF for 'algorithm' which includes 'token' as third alternative; note: if 'algorithm' was not present, MD5 is still assumed even in RFC 7616). Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd	5 years ago
Sean Bright	52ca2323aa	chan_sip.c: Don't build by default ASTERISK-29083 #close Change-Id: I9ea25fba3ba8f63a47886894bd966e0f08d5e003	5 years ago
Dennis Buteyn	aab666bb9d	chan_sip: Clear ToHost property on peer when changing to dynamic host The ToHost parameter was not cleared when a peer's host value was changed to dynamic. This causes invites to be sent to the original host. ASTERISK-29011 #close Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c	5 years ago
Frederic LE FOLL	a423f935c9	chan_sip: chan_sip does not process 400 response to an INVITE. chan_sip handle_response() function, for a 400 response to an INVITE, calls handle_response_invite() and does not generate ACK. handle_response_invite() does not recognize 400 response and has no default response processing for unexpected responses, thus it does not generate ACK either. The ACK on response repetition comes from handle_response() mechanism "We must re-send ACKs to re-transmitted final responses". According to code history, 400 response specific processing was introduced with commit "channels/chan_sip: Add improved support for 4xx error codes" This commit added support for : - 400/414/493 in handle_response_subscribe() handle_response_register() and handle_response(). - 414/493 only in handle_response_invite(). This fix adds 400 response support in handle_response_invite(). ASTERISK-28957 Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad	5 years ago
Alexander Traud	52f07176b6	chan_sip: externhost/externaddr with non-default TCP/TLS ports. ASTERISK-28372 Reported by: Anton Satskiy ASTERISK-24428 Reported by: sstream Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c	5 years ago
Alexander Traud	4d0ab620be	chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets. ASTERISK-27195 Reported by: Joshua Roys Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de	5 years ago
traud	da9554d925	chan_sip: TCP/TLS client without server. It is possible to configure a TCP/TLS client without having a TCP/TLS server. In that case, no error or warning was printed but the headers Contact and Via in SIP REGISTER were "(null)". ASTERISK-28798 Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2	5 years ago
Joshua C. Colp	1b6c58896f	chan_sip: Send 403 when ACL fails. Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2	5 years ago
Walter Doekes	43620cbf6c	chan_sip: Return 503 if we're out of RTP ports If you're for some reason out of RTP ports, chan_sip would previously responde to an INVITE with a 403, which will fail the call. Now, it returns a 503, allowing the device/proxy to retry the call on a different machine. ASTERISK-28718 Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90	5 years ago
Walter Doekes	711a3fed56	chan_sip: Always process updated SDP on media source change Fixes no-audio issues when the media source is changed and strictrtp is enabled (default). If the peer media source changes, the SDP session version also changes. If it is lower than the one we had stored, chan_sip would ignore it. This changeset keeps track of the remote media origin identifier, comparing that as well. If it changes, the session version needn't be higher for us to accept the SDP. Common scenario where this would've caused problems: a separate media gateway that informs the caller about premium rates before handing off the call to the final destination. (An alternative fix would be to set ignoresdpversion=yes on the peer.) ASTERISK-28686 Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee	5 years ago
Sean Bright	f309b86e36	chan_sip.c: Stop handling continuation lines after reading headers lws2sws() does not stop trying to handle header continuation lines even after all headers have been found. This is problematic if the first character of a SIP message body is a space or tab character, so we update to recognize the end of the message header. ASTERISK-28693 #close Reported by: Frank Matano Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df	5 years ago
Friendly Automation	2a8f759374	Merge "chan_sip: voice frames are no longer transmitted after emitting a COLP"	6 years ago
Friendly Automation	c3cf0e330c	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors."	6 years ago
Jaco Kroon	365d007eb6	chan_sip: in case of tcp/tls, be less annoying about tx errors. chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to 213.150.203.60:1492 returned -2: Interrupted system call returned -2 implies this wasn't actually an OS error, so errno makes no sense either. Internal error was already logged higher up, and -2 generally means that either there isn't a valid connection available, or the pipe notification failed, and that is already correctly logged. ASTERISK-28651 #close Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Jean Aunis	9c9296c635	chan_sip: voice frames are no longer transmitted after emitting a COLP The SIP transaction state was reset when emitting an UPDATE or a re-INVITE related to a COLP, preventing RTP packets to be emitted. ASTERISK-28647 Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b	6 years ago
Frederic LE FOLL	7624cbb155	chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime. During capabilities selection (joint capabilities of us and peer, configured capability for this peer, or general configured capabilities), if sip_new() does not keep framing information, then directmedia activation will fail for any framing different from default framing. ASTERISK-28637 Change-Id: I99257502788653c2816fc991cac7946453082466	6 years ago
Ben Ford	4a1cadeadb	chan_sip.c: Prevent address change on unauthenticated SIP request. If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option. ASTERISK-28589 #close Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df	6 years ago
Sean Bright	32ce6e9a06	channels: Allow updating variable value When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	6 years ago
George Joseph	5fb9b23105	chan_sip: Update links referenced in deprecation notice The links in the deprecation notice were the shortened variety but it makes better sense to show the unshortened links as they're more descriptive. I.E. wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip rather than wiki.asterisk.org/wiki/x/tAHOAQ Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9	6 years ago
Kevin Harwell	857ee76f4b	Merge "MWI: Update modules that subscribe to MWI to use new API calls"	6 years ago
Francesco Castellano	8438d19b81	chan_sip: Handle invalid SDP answer to T.38 re-invite The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87	6 years ago
Kevin Harwell	9637e1dfdc	MWI: Update modules that subscribe to MWI to use new API calls The MWI core recently got some new API calls that make tracking MWI state lifetime more reliable. This patch updates those modules that subscribe to specific MWI topics to use the new API. Specifically, these modules now subscribe to both MWI topics and MWI state. ASTERISK-28442 Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd	6 years ago
Kevin Harwell	ff0d0ac23a	mwi core: Move core MWI functionality into its own files There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	6 years ago
Ben Ford	dd1cc7791c	build: Fix compiler warnings/errors. The compiler complained about a couple of variables that weren't initialized but were being used. Initializing them to NULL resolves the warnings/errors. ASTERISK-28362 #close Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b	6 years ago
Sean Bright	1499640da9	chan_sip: Ensure 'qualifygap' isn't negative Passing negative intervals to the scheduler rips a hole in the space-time continuum. ASTERISK-25792 #close Reported by: Paul Sandys Change-Id: Ie706f21cee05f76ffb6f7d89e9c867930ee7bcd7	6 years ago
Sean Bright	2473b791b9	Replace calls to strtok() with strtok_r() strtok() uses a static buffer, making it not thread safe. Also add a #define to cause a compile failure if strtok is used. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	6 years ago
Giuseppe Sucameli	0bde3751a0	chan_sip: Fix leak using contact ACL Free old peer's contactacl before overwrite it within build_peer. ASTERISK-28194 Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c	7 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	3077ad0c24	stasis: Add internal filtering of messages. This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
Jasper Hafkenscheid	2cf5079205	chan_sip: Attempt ast_do_pickup in handle_invite_replaces When a call pickup is performed using and invite with replaces header the ast_do_pickup method is attempted and a PICKUP stasis message is sent. ASTERISK-28081 #close Reported-by: Luit van Drongelen Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a	7 years ago
Corey Farrell	90a11c4ae7	chan_sip deprecation. This officially deprecates chan_sip in Asterisk 17+. A warning is printed upon startup or module load to tell users that they should consider migrating. chan_sip is still built by default but the default modules.conf skips loading it at startup. Very important to note we are not scheduling a time where chan_sip will be removed. The goal of this change is to accurately inform end users of the current state of chan_sip and encourage movement to the fully supported chan_pjsip. Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93	7 years ago
Corey Farrell	687ab7aeee	astobj2: Eliminate legacy container allocation macros. These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are also removed. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
pk16208	6627c56b3d	chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI With tls and udp enabled asterisk generates a warning about sending message via udp instead of tls. sip notify command via cli works as expected and without warning. asterisk has to set the connection information accordingly to connection and not on presumption ASTERISK-28057 #close Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e	7 years ago
hajekd	406be41f21	chan_sip.c: chan_sip unstable with TLS after asterisk start or reloads Fixes random asterisk crash on start or reload with TLS phones. ASTERISK-28034 #close Reported-by: David Hajek Change-Id: I2a859f97dc80c348e2fa56e918214ee29521c4ac	7 years ago
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

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