226 Commits (6415562a140313dbcd5fd33e0991b0c1081ac4e2)

Author	SHA1	Message	Date
Ben Ford	aba7168abe	res_pjsip_session: Added new function calls to avoid ABI issues. ... Added two new functions (ast_sip_session_get_dialog and ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the pjsip_inv_state respectively from the pjsip_inv_session on the ast_sip_session struct. This is due to pjproject adding a new field to the pjsip_inv_session struct that caused crashes when trying to access fields that were no longer where they were expected to be if a module was compiled against a different version of pjproject. Resolves: #145	2 years ago
George Joseph	4afe2355be	res_pjsip: Add utils for checking media types ... Added two new functions to assist checking media types... * ast_sip_are_media_types_equal compares two pjsip_media_types. * ast_sip_is_media_type_in tests if one media type is in a list of others. Added static definitions for commonly used media types to res_pjsip.h. Changed several modules to use the new functions and static definitions. ASTERISK_29813 (not ready to close) Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9	3 years ago
Alexander Traud	bae495601a	res_pjsip: Fix for Doxygen. ... ASTERISK-29747 Change-Id: Ic7a1e9453f805a6264fe86c96b7d18b87b376084	4 years ago
Josh Soref	dcf492e7b6	res: Spelling fixes ... Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda	4 years ago
Ben Ford	2e55c0fded	STIR/SHAKEN: Option split and response codes. ... The stir_shaken configuration option now has 4 different choices to pick from: off, attest, verify, and on. Off and on behave the same way they do now. Attest will only perform attestation on the endpoint, and verify will only perform verification on the endpoint. Certain responses are required to be sent based on certain conditions for STIR/SHAKEN. For example, if we get a Date header that is outside of the time range that is considered valid, a 403 Stale Date response should be sent. This and several other responses have been added. Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7	4 years ago
Joshua C. Colp	523a795289	AST-2021-007 - res_pjsip_session: Don't offer if no channel exists. ... If a re-INVITE is received after we have sent a BYE request then it is possible for no channel to be present on the session. If this occurs we allow PJSIP to produce the offer instead. Since the call is being hung up if it produces an incorrect offer it doesn't actually matter. This also ensures that code which produces SDP does not need to handle if a channel is not present. ASTERISK-29381 Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042	4 years ago
George Joseph	c78d0ce429	res_pjsip_session: Make reschedule_reinvite check for NULL topologies ... When the check for equal topologies was added to reschedule_reinvite() it was assumed that both the pending and active media states would actually have non-NULL topologies. We since discovered this isn't the case. We now only test for equal topologies if both media states have non-NULL topologies. The logic had to be rearranged a bit to make sure that we cloned the media states if their topologies were non-NULL but weren't equal. ASTERISK-29215 Change-Id: I61313cca7fc571144338aac826091791b87b6e17	5 years ago
Ben Ford	83b0f5963f	res_pjsip_session.c: Check topology on re-invite. ... Removes an unnecessary check for the conditional that compares the stream topologies to see if they are equal to suppress re-invites. This was a problem when a Digium phone received an INVITE that offered codecs different than what it supported, causing Asterisk to send the re-invite. ASTERISK-29303 Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63	5 years ago
Joshua C. Colp	1af2a84c8b	res_pjsip_session: Always produce offer on re-INVITE without SDP. ... When PJSIP receives a re-INVITE without an SDP offer the INVITE session library will first call the on_create_offer callback and if unavailable then use the active negotiated SDP as the offer. In some cases this would result in a different SDP then was previously used without an incremented SDP version number. The two known cases are: 1. Sending an initial INVITE with a set of codecs and having the remote side answer with a subset. The active negotiated SDP would have the pruned list but would not have an incremented SDP version number. 2. Using re-INVITE for unhold. We would modify the active negotiated SDP but would not increment the SDP version. To solve these, and potential other unknown cases, the on_create_offer callback has now been implemented which produces a fresh offer with incremented SDP version number. This better fits within the model provided by the INVITE session library. ASTERISK-28452 Change-Id: I2d81048d54edcb80fe38fdbb954a86f0a58281a1	5 years ago
Kevin Harwell	fad0cf12e6	AST-2021-002: Remote crash possible when negotiating T.38 ... When an endpoint requests to re-negotiate for fax and the incoming re-invite is received prior to Asterisk sending out the 200 OK for the initial invite the re-invite gets delayed. When Asterisk does finally send the re-inivite the SDP includes streams for both audio and T.38. This happens because when the pending topology and active topologies differ (pending stream is not in the active) in the delayed scenario the pending stream is appended to the active topology. However, in the fax case the pending stream should replace the active. This patch makes it so when a delay occurs during fax negotiation, to or from, the audio stream is replaced by the T.38 stream, or vice versa instead of being appended. Further when Asterisk sent the re-invite with both audio and T.38, and the endpoint responded with a declined T.38 stream then Asterisk would crash when attempting to change the T.38 state. This patch also puts in a check that ensures the media state has a valid fax session (associated udptl object) before changing the T.38 state internally. ASTERISK-29203 #close Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09	5 years ago
Robert Cripps	017e09b40a	res/res_pjsip_session.c: Check that media type matches in ... function ast_sip_session_media_state_add. Check ast_media_type matches when a ast_sip_session_media is found otherwise when transitioning from say image to audio, the wrong session is returned in the first if statement. ASTERISK-29220 #close Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d	5 years ago
Alexander Traud	3f119192bb	res_pjsip_session: Avoid sometimes-uninitialized warning with Clang. ... ASTERISK-29248 Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556	5 years ago
Ivan Poddubnyi	c3fad2fd01	chan_pjsip: Assign SIPDOMAIN after creating a channel ... session->channel doesn't exist until chan_pjsip creates it, so intead of setting a channel variable every new incoming call sets one and the same global variable. This patch moves the code to chan_pjsip so that SIPDOMAIN is set on a newly created channel, it also removes a misleading reference to channel->session used to fetch call pickup configuraion. ASTERISK-29240 Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755	5 years ago
Richard Mudgett	027f4e3a21	res_pjsip_session.c: Fix compiler warnings. ... AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an unsigned long on all machines. Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338	5 years ago
Sungtae Kim	d8b7a6f599	res_pjsip_session: Fixed NULL active media topology handle ... Added NULL pointer check to prevent Asterisk crash. ASTERISK-29215 Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95	5 years ago
Joshua C. Colp	5b4e71fa0a	pjsip: Match lifetime of INVITE session to our session. ... In some circumstances it was possible for an INVITE session to be destroyed while we were still using it. This occurred due to the reference on the INVITE session being released internally as a result of its state changing to DISCONNECTED. This change adds a reference to the INVITE session which is released when our own session is destroyed, ensuring that the INVITE session remains valid for the lifetime of our session. ASTERISK-29022 Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9	5 years ago
Kevin Harwell	6baa4b53be	AST-2020-001 - res_pjsip: Return dialog locked and referenced ... pjproject returns the dialog locked and with a reference. However, in Asterisk the method that handles this decrements the reference and removes the lock prior to returning. This makes it possible, under some circumstances, for another thread to free said dialog before the thread that created it attempts to use it again. Of course when the thread that created it tries to use a freed dialog a crash can occur. This patch makes it so Asterisk now returns the newly created dialog both locked, and with an added reference. This allows the caller to de-reference, and unlock the dialog when it is safe to do so. In the case of a new SIP Invite the lock, and reference are now held for the entirety of the new invite handling process. Otherwise it's possible for the dialog, or its dependent objects, like the transaction, to disappear. For example if there is a TCP transport error. ASTERISK-29057 #close Change-Id: I5ef645a47829596f402cf383dc02c629c618969e	5 years ago
Ben Ford	82325ba58b	AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. ... If Asterisk sends out and INVITE and receives a challenge with a different nonce value each time, it will continually send out INVITEs, even if the call is hung up. The endpoint must be configured for outbound authentication in order for this to occur. A limit has been set on outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. ASTERISK-29013 Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7	5 years ago
Kevin Harwell	439f7bb848	res_pjsip, res_pjsip_session: initialize local variables ... This patch initializes a couple of local variables to some default values. Interestingly, in the 'pj_status_t dlg_status' case the value not being initialized caused memory to grow, and not be recovered, in the off nominal path (at least on my machine). Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7	5 years ago
Nick French	f041763e3b	res_pjsip_session: Restore calls to ast_sip_message_apply_transport() ... Commit 44bb0858cb ("debugging: Add enough to choke a mule") accidentally removed calls to ast_sip_message_apply_transport when it was attempting to just add debugging code. The kiss of death was saying that there were no functional changes in the commit comment. This makes outbound calls that use the 'flow' transport mechanism fail, since this call is used to relay headers into the outbound INVITE requests. ASTERISK-29124 #close Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a	5 years ago
Joshua C. Colp	9eeb40af33	res_pjsip_session: Fix stream name memory leak. ... When constructing a stream name based on the media type and position the allocated name was not being freed causing a leak. Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de	5 years ago
Joshua C. Colp	957aff751d	res_pjsip_session: Fix session reference leak. ... The ast_sip_dialog_get_session function returns the session with reference count increased. This was not taken into account and was causing sessions to remain around when they should not be. ASTERISK-29089 Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8	5 years ago
Sean Bright	6f32c254be	res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined ... Change-Id: Id4852c26e9c412af8e37b5dd3c15da9453ad3276	5 years ago
George Joseph	cc71be0078	res_pjsip_session: Fix issue with COLP and 491 ... The recent 491 changes introduced a check to determine if the active and pending topologies were equal and to suppress the re-invite if they were. When a re-invite is sent for a COLP-only change, the pending topology is NULL so that check doesn't happen and the re-invite is correctly sent. Of course, sending the re-invite sets the pending topology. If a 491 is received, when we resend the re-invite, the pending topology is set and since we didn't request a change to the topology in the first place, pending and active topologies are equal so the topologies-equal check causes the re-invite to be erroneously suppressed. This change checks if the topologies are equal before we run the media state resolver (which recreates the pending topology) so that when we do the final topologies-equal check we know if this was a topology change request. If it wasn't a change request, we don't suppress the re-invite even though the topologies are equal. ASTERISK-29014 Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314	5 years ago
George Joseph	ad4f2a8c99	debugging: Add enough to choke a mule ... Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	5 years ago
George Joseph	d4f3b17dd3	res_pjsip_session: Handle multi-stream re-invites better ... When both Asterisk and a UA send re-invites at the same time, both send 491 "Transaction in progress" responses to each other and back off a specified amount of time before retrying. When Asterisk prepares to send its re-invite, it sets up the session's pending media state with the new topology it wants, then sends the re-invite. Unfortunately, when it received the re-invite from the UA, it partially processed the media in the re-invite and reset the pending media state before sending the 491 losing the state it set in its own re-invite. Asterisk also was not tracking re-invites received while an existing re-invite was queued resulting in sending stale SDP with missing or duplicated streams, or no re-invite at all because we erroneously determined that a re-invite wasn't needed. There was also an issue in bridge_softmix where we were using a stream from the wrong topology to determine if a stream was added. This also caused us to erroneously determine that a re-invite wasn't needed. Regardless of how the delayed re-invite was triggered, we need to reconcile the topology that was active at the time the delayed request was queued, the pending topology of the queued request, and the topology currently active on the session. To do this we need a topology resolver AND we need to make stream named unique so we can accurately tell what a stream has been added or removed and if we can re-use a slot in the topology. Summary of changes: * bridge_softmix: * We no longer reset the stream name to "removed" in remove_all_original_streams(). That was causing multiple streams to have the same name and wrecked the checks for duplicate streams. * softmix_bridge_stream_sources_update() was checking the old_stream to see if it had the softmix prefix and not considering the stream as "new" if it did. If the stream in that slot has something in it because another re-invite happened, then that slot in old might have a softmix stream but the same stream in new might actually be a new one. Now we check the new_stream's name instead of the old_stream's. * stream: * Instead of using plain media type name ("audio", "video", etc) as the default stream name, we now append the stream position to it to make it unique. We need to do this so we can distinguish multiple streams of the same type from each other. * When we set a stream's state to REMOVED, we no longer reset its name to "removed" or destroy its metadata. Again, we need to do this so we can distinguish multiple streams of the same type from each other. * res_pjsip_session: * Added resolve_refresh_media_states() that takes in 3 media states and creates an up-to-date pending media state that includes the changes that might have happened while a delayed session refresh was in the delayed queue. * Added is_media_state_valid() that checks the consistency of a media state and returns a true/false value. A valid state has: * The same number of stream entries as media session entries. Some media session entries can be NULL however. * No duplicate streams. * A valid stream for each non-NULL media session. * A stream that matches each media session's stream_num and media type. * Updated handle_incoming_sdp() to set the stream name to include the stream position number in the name to make it unique. * Updated the ast_sip_session_delayed_request structure to include both the pending and active media states and updated the associated delay functions to process them. * Updated sip_session_refresh() to accept both the pending and active media states that were in effect when the request was originally queued and to pass them on should the request need to be delayed again. * Updated sip_session_refresh() to call resolve_refresh_media_states() and substitute its results for the pending state passed in. * Updated sip_session_refresh() with additional debugging. * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE to pjproject if a transaction is in progress. This stops us from creating a partial pending media state that would be invalid later on. * Updated reschedule_reinvite() to clone both the current pending and active media states and pass them to delay_request() so the resolver can tell what the original intention of the re-invite was. * Added a large unit test for the resolver. ASTERISK-29014 Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb	5 years ago
Patrick Verzele	5a49757e40	res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly ... Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again. Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6	5 years ago
Joshua C. Colp	3c074038fe	res_pjsip_session: Don't aggressively terminate on failed re-INVITE. ... Per the RFC when an outgoing re-INVITE is done we should only terminate the dialog if a 481 or 408 is received. ASTERISK-29033 Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503	5 years ago
George Joseph	71446b68fc	res_pjsip_session: Ensure reused streams have correct bundle group ... When a bundled stream is removed, its bundle_group is reset to -1. If that stream is later reused, the bundle parameters on session media need to be reset correctly it could mistakenly be rebundled with a stream that was removed and never reused. Since the removed stream has no rtp instance, a crash will result. Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7	5 years ago
George Joseph	e88beedd08	res_pjsip_session: Fix segv in session_on_rx_response ... session_on_rx_response wasn't checking for a NULL dialog before attempting to get the invite session from it. Change-Id: Id13534375966cc2eb7f2b55717c9813c63c10065	5 years ago
George Joseph	9bd1d686a1	ACN: Add tracing to existing code ... Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	5 years ago
Kevin Harwell	4eba6b9eb2	PJSIP_MEDIA_OFFER: override configuration on refresh ... When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs on refresh unless they had a shared codec between the two. This patch makes it so whatever is set using PJSIP_MEDIA_OFFER is used when creating the SDP for a refresh no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	5 years ago
Joshua C. Colp	ee8ea9275f	res_pjsip_session: Preserve label on incoming re-INVITE. ... When a re-INVITE is received we create a new set of streams that are then swapped in as the active streams. We did not preserve the SDP label from the previous streams, resulting in the label getting lost. This change ensures that if an SDP label is present on the previous stream then it is set on the new stream. ASTERISK-28953 Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445	5 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
Joshua C. Colp	e56f4de7e6	fax: Fix crashes in PJSIP re-negotiation scenarios. ... This change fixes a few re-negotiation issues uncovered with fax. 1. The fax support uses its own mechanism for re-negotiation by conveying T.38 information in its own frames. The new support for re-negotiating when adding/removing/changing streams was also being triggered for this causing multiple re-INVITEs. The new support will no longer trigger when transitioning between fax. 2. In off-nominal re-negotiation cases it was possible for some state information to be left over and used by the next re-negotiation. This is now cleared. ASTERISK-28811 ASTERISK-28839 Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2	6 years ago
DanielYK	9f117ac9ef	res_pjsip: Fixed format of IPv6 addresses for external media addresses ... ASTERISK-28835 Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26	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
Torrey Searle	e12244153a	res_pjsip_session: implement processing of Content-Disposition ... RFC5621 requires any content type with a Content-Disposition with handling=required to be rejected with a 415 response ASTERISK-28782 #close Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4	6 years ago
Joshua C. Colp	21e9051461	res_pjsip_session: Apply intention behind requested formats. ... When an outgoing channel is created a list of formats may optionally be provided which is used as a request that the formats be used if possible. If an endpoint is not configured for any of the formats we ignore this request and use what is configured. This has the side effect of also including other stream types (such as video) that were not present in the requested formats. This change makes it so that the intention of the request is preserved - that is if only an audio format is requested then even if there is no joint audio format between the request and the configuration we will still only place an audio stream in the outgoing call. ASTERISK-28787 Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f	6 years ago
Sungtae Kim	8147f43756	res_pjsip_session: Fixed wrong session termination ... When the Asterisk receives 200 OK with invalid SDP, the Asterisk/PJPROJECT terminating the session. But if the channel was in the Bridge, Asterisk tries send the Re-Invite before terminating the session. And when the Asterisk sending the Re-Invite, it doesn't check the SDP is NULL or not. This crashes the Asterisk. Fixed it to close the session correctly if the UAS sends the 200 OK with wrong SDP. ASTERISK-28743 Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da	6 years ago
Joshua C. Colp	9620ecbf80	res_pjsip_session: Don't restrict non-audio default streams to sendrecv. ... The state of the default audio stream is used for hold/unhold so we restrict it to sendrecv as the core does not handle when it changes as a result of hold/unhold. This restriction does not apply to other media types though so we now only restrict it to audio. This allows the other default streams to store their state at all values, and not just sendrecv and removed. ASTERISK-28783 Change-Id: I139740f38cea7f7d92a876ec2631ef50681f6625	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	5a5be92b79	bridging: Add better support for adding/removing streams. ... This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	6 years ago
Joshua C. Colp	ac155decae	res_pjsip_session: Fix off-nominal session refreshes. ... Given a scenario where session refreshes occur close to each other while another is finishing it was possible for the session refreshes to occur out of order. It was also possible for session refreshes to be delayed for quite some time if a session refresh did not result in a topology change. For the out of order session refreshes the first session refresh would be queued due to a transaction in progress. This transaction would then finish. When finished a separate task to process the delayed requests queue would be queued for handling. A second refresh would be requested internally before this delayed request queued task was processed. As no transaction was in progress this session refresh would be immediately handled before the queued session refresh. The code will now check if any delayed requests exist before allowing a session refresh to immediately occur. If any exist then the session refresh is queued. For the delayed session refreshes if a session refresh did not result in a topology change the attempt would be immediately stopped and no other delayed requests would be processed. The code will now go through the entire delayed requests queue until a delayed request results in a request actually being sent. ASTERISK-28730 Change-Id: Ied640280133871f77d3f332be62265e754605088	6 years ago
Joshua C. Colp	a603d7d324	res_pjsip_session: Set stream state on created streams for incoming SDP. ... A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355	6 years ago
Sean Bright	6ee1f1f507	res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled ... We need to copy the endpoint name before we call ao2_cleanup() on it, otherwise we might try to access memory that has been reclaimed. ASTERISK-28445 #close Reported by: Bernhard Schmidt Change-Id: I404b952608aa606e0babd3c4108346721fb726b3	6 years ago
Kevin Harwell	0e3b397812	res_pjsip_session: initialize pending's topology to endpoint's ... Found during some testing, there is a race condition between selecting an appropriate bridge type for a call versus the applying of media on the callee's session. In some instances a native bridge type would have been chosen, but due to the callee's media not yet being established at bridge compatibility check time the simple bridge type is picked instead. When using chan_pjsip this initiates a topology change event. The topologies are then compared for the two sessions. However, when the topology was created for the caller its streams are initialized to "inactive". This topology is then used as a base when creating the callee's topology, and streams. Soon after the caller's topology's stream(s) get updated based on the sdp (get set to sendrecv in the failing scenario). Now when the topology change event is raised, and the two topologies are compared, the comparison fails due to a stream state mismatch (sendrecv vs inactive). And since they differ a reinvite is sent out (to the caller in this case). This patch makes it such that when the caller's topology is initially created it gets created based on its configured endpoint's media topology. When the endpoint's topology is created its stream's state(s) are initialized to sendrecv instead of inactive. Subsequently, now when the callee's topology is created its topology streams are now initialized to sendrecv. Thus when the topology change event occurs due to the mentioned scenario the stream states match for the given sessions, and the reinvite is not sent unless due to some other valid mismatch. Note, this patch only changes one pending media state's creation point. It's possible other places *could* be changed, however for now it was deemed best to only alter what's here. Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7	6 years ago
Torrey Searle	b43cdc7f1e	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	6 years ago
Stas Kobzar	c7270dca81	res_pjsip: Channel variable SIPFROMDOMAIN ... In chan_sip, there was variable SIPFROMDOMAIN that allows to set From header URI domain per channel. This patch introduces res_pjsip variable SIPFROMDOMAIN for backward compatibility with chan_sip. ASTERISK-28489 Change-Id: I715133e43172ce2a1e82093538dc39f9e99e5f2e	6 years ago
Sungtae Kim	7e1d881d89	res_pjsip_session Added rtcp stats result vector into the session ... Currently, the Asterisk's pjsip_session module does not keeping the rtcp's stats info after it was removed. But by adding the results vector and keeping it until session is destroying, it can give more useful information for other modules. ASTERISK-28253 Change-Id: Ib25c2d3fc4da084aecfde2a82c1b1d733bd64fa5	7 years ago