459 Commits (1c5e68580af4556de86c2c33200fe3d43d499bed)

Author	SHA1	Message	Date
Joshua C. Colp	d845464c76	res_pjsip: Don't set endpoint to unavailable in all cases. ... When an AOR is modified endpoints are updated that reference the AOR so they can start receiving updates and reflect the correct state. If this is the case then we shouldn't change the endpoint to be offline if it does not reference the AOR but instead only when the endpoint is completely updated for all its AORs. ASTERISK-28056 patches: pjsip_options-aor.diff submitted by jhord (license 6978) Change-Id: I3ee00023be2393113cd4e056599f23f3499ef164	6 years ago
George Joseph	7ba6d43083	test_res_pjsip_session_caps: Create unit test ... This unit test runs through combinations of... * Local codecs * Remote Codecs * Codec Preference * Incoming/Outgoing A few new APIs were created to make it easier to test the functionality but didn't result in any actual functional change. ASTERISK_28777 Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164	6 years ago
George Joseph	2ee455958e	codec_negotiation: Implement outgoing_call_offer_pref ... Based on this new endpoint setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) * Add "call_direction" to res_pjsip_session. * Update pjsip_session_caps.c to make the functions more generic so they could be used for both incoming and outgoing. * Update ast_sip_session_create_outgoing to create the pending_media_state->topology with the results of ast_sip_session_create_joint_call_stream(). * The endpoint "preferred_codec_only" option now automatically sets AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. * A helper function ast_stream_get_format_count() was added to streams to return the current count of formats. ASTERISK-28777 Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437	6 years ago
Kevin Harwell	06dada3f01	codec negotiation: add incoming_call_offer_prefs option ... Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that specifies the preferred order of codecs after receiving an offer. This patch does the following: Adds a new enumeration, ast_sip_call_codec_pref, used by the the new configuration option that's added to the endpoint media structure. Adds a new ast_sip_session_caps structure that's set for each session media object. Creates a new file, res_pjsip_session_caps that "implements" the new structure and option, and is compiled into the res_pjsip_session library. ASTERISK-28756 #close Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f	6 years ago
Joshua C. Colp	d6712790cd	pjsip: Update ACLs on named ACL changes. ... This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	6 years ago
George Joseph	cbc1136704	res_pjsip_nat: Restore original contact for REGISTER responses ... RFC3261 Section 10 "Registrations", specifically paragraph "10.2.4: Refreshing Bindings", states that a user agent compares each contact address (in a 200 REGISTER response) to see if it created the contact. If the Asterisk endpoint has the rewrite_contact option set however, the contact host and port sent back in the 200 response will be the rewritten one and not the one sent by the user agent. This prevents the user agent from matching its own contact. Some user agents get very upset when this happens and will not consider the registration successful. While this is rare, it is acceptable behavior especially if more than 1 user agent is allowed to register to a single endpoint/aor. This commit updates res_pjsip_nat (where rewrite_contact is implemented) to store the original incoming Contact header in a new "x-ast-orig-host" URI parameter before rewriting it, and to restore the original host and port to the Contact headers in the outgoing response. This is only done if the request is a REGISTER and rewrite_contact is enabled. pjsip_message_filter was also updated to ensure that if a request comes in with any existing x-ast-* URI parameters, we remove them so they don't conflict. Asterisk will never send a request with those headers in it but someone might just decide to add them to a request they craft and send to Asterisk. NOTE: If a device changes its contact address and registers again, it's a NEW registration. If the device didn't unregister the original registration then all existing behavior based on aor/remove_existing and aor/max_contacts apply. ASTERISK-28502 Reported-by: Ross Beer Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e	6 years ago
Kevin Harwell	bdd785d31c	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	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
Joshua Colp	86452c9fa4	res_pjsip: Fix multiple of the same contact in "pjsip show contacts". ... The code for gathering contacts could result in the same contact being retrieved and added to the list multiple times. The container which stores the contacts to display will now only allow a contact to be added to it once instead of multiple times. ASTERISK-28228 Change-Id: I805185cfcec03340f57d2b9e6cc43c49401812df	6 years ago
Ben Ford	86836e0442	pjsip_options.c: Allow immediate qualifies for new contacts. ... When multiple endpoints try to register close together using the same AOR with qualify_frequency set, one contact would qualify immediately while the other contacts would have to wait out the duration of the timer before being able to qualify. Changing the conditional to check the contact container count for a non-zero value allows all contacts to qualify immediately. Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415	7 years ago
Joshua Colp	9c070a4ae3	Merge "res_pjsip: Added a norefersub configuration setting"	7 years ago
Dan Cropp	cffa2a74cb	res_pjsip: Added a norefersub configuration setting ... Added a new PJSIP global setting called norefersub. Default is true to keep support working as before. res_pjsip_refer: Configures PJSIP norefersub capability accordingly. Checks the PJSIP global setting value. If it is true (default) it adds the norefersub capability to PJSIP. If it is false (disabled) it does not add the norefersub capability to PJSIP. This is useful for Cisco switches that do not follow RFC4488. ASTERISK-28375 #close Reported-by: Dan Cropp Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9	7 years ago
Alexei Gradinari	fe58bc7bdf	res_pjsip: Fix transport_states ref leak ... Add missing ao2_ref(transport_state, -1) while iterate on a transport_states container. Change-Id: I40e35b5a339121300c80075c30db47201a6c374e	7 years ago
Torrey Searle	4661c08549	chan_pjsip: add a flag to ignore 183 responses if no SDP present ... chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	7 years ago
Joshua Colp	c20b84e35b	Merge "res_pjsip_registrar: blocked threads on reliable transport shutdown take 3"	7 years ago
Sean Bright	101272d0dc	Revert "pjsip_message_filter: Only do interface lookup for wildcard addresses." ... This reverts commit d524ad523d. Reason for revert: This causes Contact and Via headers to have the wrong transport address. ASTERISK-28309 #close Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8	7 years ago
Kevin Harwell	930a7fe910	res_pjsip_registrar: blocked threads on reliable transport shutdown take 3 ... When a contact was removed by the registrar it did not always check to see if the circumstances involved a monitored reliable transport. For instance, if the 'remove_existing' option was set to 'true' then when existing contacts were removed due to 'max_contacts' being reached, those existing contacts being removed did not unregister the transport monitor. Also, it was possible to add more than one monitor on a reliable transport for a given aor and contact. This patch makes it so all contact removals done by the registrar also remove any associated transport monitors if necessary. It also makes it so duplicate monitors cannot be added for a given transport. ASTERISK-28213 Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a	7 years ago
George Joseph	c2adeb9dc2	taskprocessor: Enable subsystems and overload by subsystem ... To prevent one subsystem's taskprocessors from causing others to stall, new capabilities have been added to taskprocessors. * Any taskprocessor name that has a '/' will have the part before the '/' saved as its "subsystem". Examples: "sorcery/acl-0000006a" and "sorcery/aor-00000019" will be grouped to subsystem "sorcery". "pjsip/distributor-00000025" and "pjsip/distributor-00000026" will bn grouped to subsystem "pjsip". Taskprocessors with no '/' have an empty subsystem. * When a taskprocessor enters high-water alert status and it has a non-empty subsystem, the subsystem alert count will be incremented. * When a taskprocessor leaves high-water alert status and it has a non-empty subsystem, the subsystem alert count will be decremented. * A new api ast_taskprocessor_get_subsystem_alert() has been added that returns the number of taskprocessors in alert for the subsystem. * A new CLI command "core show taskprocessor alerted subsystems" has been added. * A new unit test was addded. REMINDER: The taskprocessor code itself doesn't take any action based on high-water alerts or overloading. It's up to taskprocessor users to check and take action themselves. Currently only the pjsip distributor does this. * A new pjsip/global option "taskprocessor_overload_trigger" has been added that allows the user to select the trigger mechanism the distributor uses to pause accepting new requests. "none": Don't pause on any overload condition. "global": Pause on ANY taskprocessor overload (the default and current behavior) "pjsip_only": Pause only on pjsip taskprocessor overloads. * The core pjsip pool was renamed from "SIP" to "pjsip" so it can be properly grouped into the "pjsip" subsystem. * stasis taskprocessor names were changed to "stasis" as the subsystem. * Sorcery core taskprocessor names were changed to "sorcery" to match the object taskprocessors. Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56	7 years ago
Kevin Harwell	f668db9ba0	pjsip/config_global: regcontext context not created ... The context specified by 'regcontext' was not being created, so when Asterisk attempted to later dynamically add an extension it would fail. This patch now creates the context if a 'regcontext' is specified. ASTERISK-28238 Change-Id: I0f36cf4ab0a93ff4b1cc5548d617ecfd45e09265	7 years ago
Sean Bright	20f672539e	pjsip_transport_management: Shutdown transport immediately on disconnect ... The transport management code that checks for idle connections keeps a reference to PJSIP's transport for IDLE_TIMEOUT milliseconds (32000 by default). Because of this, if the transport is closed before this timeout, the idle checking code will keep the transport from actually being shutdown until the timeout expires. Rather than passing the AO2 object to the scheduler task, we just pass its key and look it up when it is time to potentially close the idle connection. The other transport management code handles cleaning up everything else for us. Additionally, because we use the address of the transport when generating its name, we concatenate an incrementing ID to the end of the name to guarantee uniqueness. Related to ASTERISK~28231 Change-Id: I02ee9f4073b6abca9169d30c47aa69b5e8ae9afb	7 years ago
Alexei Gradinari	f0546d1d87	res_pjsip: add option to enable ContactStatus event when contact is updated ... The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out the ContactStatus AMI event when a contact is updated. Thist change broke things which rely on old behavior. This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to preserve old ContactStatus behavior. By default new behavior, i.e. the ContactStatus event will not be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	7 years ago
Sungtae Kim	8644511cbf	res_pjsip: Patch for res_pjsip_* module load/reload crash ... The session_supplements for the pjsip makes crashes when the module load/unload. ASTERISK-28157 Change-Id: I5b82be3a75d702cf1933d8d1417f44aa10ad1029	7 years ago
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	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
Chris-Savinovich	a3fc97aa13	res_pjsip: Send a 503 response when overload state if reliable transport. ... When Asterisk's taskprocessors get overloaded we need to reduce the work load. res_pjsip currently ignores new SIP requests and relies on SIP retransmissions in the hope that the overload condition will clear soon enough to handle the retransmitted SIP request. This change adds the following code after ast_taskprocessor_alert_get() has returned TRUE: 1- identifies transport type. If non-udp then send a 503 response 2- if transport type is udp/udp6 then ignore, as before. Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836	7 years ago
Alexei Gradinari	eee935983b	pjsip: new endpoint's options to control Connected Line updates ... This patch adds new options 'trust_connected_line' and 'send_connected_line' to the endpoint. The option 'trust_connected_line' is to control if connected line updates are accepted from this endpoint. The option 'send_connected_line' is to control if connected line updates can be sent to this endpoint. The default value is 'yes' for both options. Change-Id: I16af967815efd904597ec2f033337e4333d097cd	7 years ago
Torrey Searle	cac4ccef25	res_pjsip_session: add new flag use_callerid_contact ... Add a new global flag to res_pjsip to allow the callerid to be used as the username in the contact header. This allows chan_pjsip to have the same behavour as chan_sip ASTERISK-28087 #close Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95	7 years ago
Nick French	37b2e68628	res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability ... This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58	7 years ago
Alexei Gradinari	aae5bdc22e	res_pjsip: set callerid_tag to empty string ... This patch sets the callerid_tag to empty string by default. If the callerid_tag is set to NULL then the tag does not become part of a connected line update. For example: Alice's tag is "Alice". Bob's tag is empty. Charlie's tag is "Charlie". Alice calls Bob and then does attended transfer to Charlie. When Alice hangs up the CONNECTEDLINE(tag) is "Alice" on the interception routine on the Charlie's channel, but should be empty. Ths patch also fix memory leaks if there are more then one options "callerid", "callerid_tag", "voicemail_extension" and "contact_user" in the pjsip.conf endpoint definition. Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4	7 years ago
Alexei Gradinari	8bb031abc7	res_pjsip: improve realtime performance on CLI 'pjsip show contacts' ... CLI command 'pjsip show contacts' inefficiently make a lot of DB requests. For example if there are 10k aors then asterisk requests these 10k records of aor and then does 10k requests of contact - one request per aor. Even if use 'like <pattern>' the asterisk requests all aor's and contact's records and then filters them by itself. This patch gathers contact's container by - retrieving all dynamic contacts by regex (filtered by reg_server) - retrieving all aors with permanent contacts - finally filters container by regex ASTERISK-28077 #close Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618	7 years ago
Joshua Colp	ce9a980be6	pjproject: Upgrade to 2.8. ... This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	7 years ago
Sean Bright	07cb13f75f	res_pjsip: Log IPv6 addresses correctly ... Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name store IPv6 addresses without enclosing brackets. This causes some log output to be confusing because it is difficult to separate the IPv6 address from a port specification. * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 output. * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses in brackets. * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure to also set pjsip_rx_data.pkt_info.src_addr. Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8	7 years ago
Richard Mudgett	d60411a2b4	res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch ... ASTERISK-27988 Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843	7 years ago
Joshua Colp	5320b18bfe	Merge "res_pjsip: Reduce processing when a Contact is updated."	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
Joshua Colp	a88cec6334	Merge "res_pjsip: Resolve transport management leak at shutdown."	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	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
Joshua Colp	39a8920504	Merge "res_pjsip: Make pjlib.h consistently included."	7 years ago
Alexander Traud	603d1e8d4b	pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. ... The authors of PJProject undef s_addr because of some issue in Microsoft Windows. However in Oracle Solaris, s_addr is not a structure member, but defined to map to the real structure member. Updates the patch from ASTERISK_20366 ASTERISK-27997 Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11	7 years ago
Richard Mudgett	acbb9f52b2	res_pjsip: Make pjlib.h consistently included. ... * Don't include pjlib.h twice in res_pjsip.h * Consistently use #include <> form for pjproject includes. (pjsip.h and pjlib.h) Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7	7 years ago
George Joseph	9e47a7ffca	Merge "res_pjsip: Update default keepalive interval to 90 seconds."	7 years ago
Joshua Colp	2c9757bc90	res_pjsip: Update default keepalive interval to 90 seconds. ... A change recently went in which disabled the built-in PJSIP keepalive. This defaulted to 90 seconds and kept TCP/TLS connections alive. Disabling this functionality has resulted in a behavior change of not doing keepalives by default resulting in TCP/TLS connections dropping for some people. This change makes our default keepalive interval 90 seconds to match the previous behavior and preserve it. ASTERISK-27978 Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6	7 years ago
Richard Mudgett	8a100ca52b	pjsip_resolver.c: Use replacement function ... * Use the replacement function ast_sip_push_task_wait_servant() instead of the deprecated ast_sip_push_task_synchronous(). Change-Id: I145b550ba7054640c7faa3b644e63137f505c612	7 years ago
Joshua Colp	59323121f3	res_sorcery_config: Allow configuration section to be used based on name. ... A problem I've seen countless times is a global or system section for PJSIP not getting applied. This is inevitably the result of the "type=" line missing. This change alleviates that problem. The ability to specify an explicit section name has been added to res_sorcery_config. If the configured section name matches this and there are no unknown things configured the section is taken as being for the given type. Both the PJSIP "global" and "system" types now support this so you can just name your section "global" or "system" and it will be matched and used, even without a "type=" line. ASTERISK-27972 Change-Id: Ie22723663c1ddd24f869af8c9b4c1b59e2476893	7 years ago
Joshua Colp	68c0c081f9	Merge "res_pjsip/pjsip_transport_management.c: Fix deadlock with transport keep alive."	7 years ago
George Joseph	8f42447c68	res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint ... A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1	7 years ago
Richard Mudgett	0aff1a278e	res_pjsip/pjsip_transport_management.c: Fix deadlock with transport keep alive. ... Using the keep_alive_interval option can result in a deadlock between the pjproject transport manager group lock and the monitored transports ao2 container lock. The pjproject transport manager group lock has to be superior in the locking order to the monitored transports ao2 container lock because of pjproject callbacks called when already holding the group lock. The lock inversion happens when Asterisk attempts to send a keep alive packet over the reliable transports. * Made keepalive_transport_thread() iterate over the monitored transports container rather than use the ao2_callback() method. This avoids holding the container lock when sending the keep alive packet. ASTERISK-26686 Change-Id: I5d5392a52e698bbe41a93f7d8e92bf0e61fe3951	7 years ago
Joshua Colp	de5144e751	pjsip: Clarify certificate configuration for Websocket. ... The Websocket transport uses the built-in HTTP server. As a result the TLS configuration is done in http.conf and not in pjsip.conf. This change adds a warning if this is configured in pjsip.conf and also clarifies in the sample configuration file. Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9	7 years ago
George Joseph	880fbff6b7	res_pjsip_session: Add ability to accept multiple sdp answers ... pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer	7 years ago