asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	32b57067a6	res_rtp_asterisk: Fix regression issues with DTLS client check * Since ICE candidates are used for the check and pjproject is required to use ICE, res_rtp_asterisk was failing to compile when pjproject wasn't available. The check is now wrapped with an #ifdef HAVE_PJPROJECT. * The rtp->ice_active_remote_candidates container was being used to check the address on incoming packets but that container doesn't contain peer reflexive candidates discovered during negotiation. This was causing the check to fail where it shouldn't. We now check against pjproject's real_ice->rcand array which will contain those candidates. * Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where we weren't zeroing out sin->sin_zero before returning. This was causing ast_sockaddr_cmp() to always return false when one of the inputs was converted from a pj_sockaddr, even if both inputs had the same address and port. Resolves: #500 Resolves: #503 Resolves: #505	2 years ago
George Joseph	e46cc258ab	res_rtp_asterisk.c: Check DTLS packets against ICE candidate list When ICE is in use, we can prevent a possible DOS attack by allowing DTLS protocol messages (client hello, etc) only from sources that are in the active remote candidates list. Resolves: GHSA-hxj9-xwr8-w8pq	2 years ago
zhengsh	37b1ceab2b	res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock. From the gdb information, it was found that when calling __ast_free, the size of the allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb, it is found to be 1. Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid, which is outside the protection of the rtp_instance lock. However, ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses rtp->themssrc_valid within the protection of the rtp_instance lock. This can lead to the possibility that the value of rtp->themssrc_valid used in the call to ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used within ast_rtcp_generate_report(). Resolves: asterisk#63 (cherry picked from commit `25a766f49d`)	2 years ago
Sean Bright	96d9ad51ac	doxygen: Fix doxygen errors. Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	3 years ago
George Joseph	2f5aece0c9	res_rtp_asterisk: Don't use double math to generate timestamps Rounding issues with double math were causing rtp timestamp slips in outgoing packets. We're now back to integer math and are getting no more slips. ASTERISK-30391 Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4	3 years ago
George Joseph	4710f37ef6	res_rtp_asterisk: Asterisk Media Experience Score (MES) ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	3 years ago
George Joseph	62ca063fca	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `d454801c2d`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86	3 years ago
George Joseph	d454801c2d	res_rtp_asterisk: Asterisk Media Experience Score (MES) This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	3 years ago
Sean Bright	5ac5c2b0ab	openssl: Supress deprecation warnings from OpenSSL 3.0 There is work going on to update our OpenSSL usage to avoid the deprecated functions but in the meantime make it possible to compile in devmode. Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726	4 years ago
Torrey Searle	7b15ced930	res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf When generating dtmfs, asterisk can incorrectly think packet loss occured during the dtmf generation, resulting in a jump in sequence numbers when forwarding voice frames resumes. This patch forces asterisk to re-learn the expected sequence number after each DTMF to avoid this ASTERISK-29869 #close Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853	4 years ago
Naveen Albert	d35e292ae4	res_rtp_asterisk: Fix typo in flag test/set The code currently checks to see if an RFC3389 warning flag is set, except if it is, it merely sets the flag again, the logic of which doesn't make any sense. This adjusts the if comparison to check if the flag has NOT been set, and if so, emit a notice log event and set the flag so that future frames do not cause an event to be logged. ASTERISK-29856 #close Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1	4 years ago
Mike Bradeen	59fcd1e7e2	res_rtp_asterisk: Addressing possible rtp range issues res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined that rtpstart was configured to be an odd value. Also adding a loop counter to prevent a possible infinite loop when looking for a free port. ASTERISK-27406 Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b	4 years ago
Alexander Traud	a85f2bf34d	res: Fix for Doxygen. These are the remaining issues found in /res. ASTERISK-29761 Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d	4 years ago
Josh Soref	9ae9893c63	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
Jean Aunis	6bc747b639	res_rtp_asterisk: fix memory leak Add missing reference decrement in rtp_deallocate_transport() ASTERISK-29671 Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9	4 years ago
Guido Falsi	29ad5b18f1	res_rtp_asterisk.c: Fix build failure when not building with pjproject. Some code has been added referencing symbols defined in a block protected by #ifdef HAVE_PJPROJECT. Protect those code parts in ifdef blocks too. ASTERISK-29660 Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f	4 years ago
Sebastien Duthil	6fbf55ac11	res_rtp_asterisk: Automatically refresh stunaddr from DNS This allows the STUN server to change its IP address without having to reload the res_rtp_asterisk module. The refresh of the name resolution occurs first when the module is loaded, then recurringly, slightly after the previous DNS answer TTL expires. ASTERISK-29508 #close Change-Id: I7955a046293f913ba121bbd82153b04439e3465f	4 years ago
Alexander Traud	63d27af3ca	res_rtp_asterisk: sqrt(.) requires the header math.h. ASTERISK-29616 Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900	4 years ago
Joshua C. Colp	a985e5069c	res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates. RTCP ICE candidates use a base address derived from the RTP candidate. The port on the base address was not being updated to the RTCP port. This change sets the base port to the RTCP port and all is well. ASTERISK-29433 Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040	4 years ago
Jeremy Lainé	d162789c4d	res_rtp_asterisk: make it possible to remove SOFTWARE attribute By default Asterisk reports the PJSIP version in a SOFTWARE attribute of every STUN packet it sends. This may not be desired in a production environment, and RFC5389 recommends making the use of the SOFTWARE attribute a configurable option: https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2 This patch adds a `stun_software_attribute` yes/no option to make it possible to omit the SOFTWARE attribute from STUN packets. ASTERISK-29434 Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b	5 years ago
Sean Bright	b1807d440e	res_rtp_asterisk: More robust timestamp checking We assume that a timestamp value of 0 represents an 'uninitialized' timestamp, but 0 is a valid value. Add a simple wrapper to be able to differentiate between whether the value is set or not. This also removes the fix for ASTERISK~28812 which should not be needed if we are checking the last timestamp appropriately. ASTERISK-29030 #close Change-Id: Ie70d657d580d9a1f2877e25a6ef161c5ad761cf7	5 years ago
Kevin Harwell	0fc906a5e1	res_rtp_asterisk: Fix standard deviation calculation For some input to the standard deviation algorithm extremely large, and wrong numbers were being calculated. This patch uses a new formula for correctly calculating both the running mean and standard deviation for the given inputs. ASTERISK-29364 #close Change-Id: Ibc6e18be41c28bed3fde06d612607acc3fbd621f	5 years ago
Kevin Harwell	c4a376aac2	res_rtp_asterisk: Don't count 0 as a minimum lost packets The calculated minimum lost packets represents the lowest number of lost packets missed during an RTCP report interval. Zero of course is the lowest, but the idea is that this value contain the lowest number of lost packets once some have been missed. This patch checks to make sure the number of lost packets over an interval is not zero before checking and setting the minimum value. Also, this patch updates the rtp lost packet test to check for packet loss over several reports vs one. Change-Id: I07d6e21cec61e289c2326138d6bcbcb3c3d5e008	5 years ago
Kevin Harwell	65b68fd060	res_rtp_asterisk: Statically declare rtp_drop_packets_data object This patch makes the drop_packets_data object static. Change-Id: If4f9b21fa0c47d41a35b6b05941d978efb4da87b	5 years ago
Joshua C. Colp	8bd13a995a	res_rtp_asterisk: Only raise flash control frame on end. Flash in RTP is conveyed the same as DTMF, just with a specific digit. In Asterisk however we do flash as a single control frame. This change makes it so that only on end do we provide the flash control frame to the core. Previously we would provide a flash control frame on both begin and end, causing flash to work improperly. ASTERISK-29373 Change-Id: I1accd9c6e859811336e670e698bd8bd124f33226	5 years ago
Kevin Harwell	b86f1ef54c	res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command This patch makes it so when Asterisk is compiled in DEVMODE a CLI command is available that allows someone to drop incoming RTP packets. The command allows for dropping of packets once, or on a timed interval (e.g. drop 10 packets every 5 seconds). A user can also specify to drop packets by IP address. Change-Id: I25fa7ae9bad6ed68e273bbcccf0ee51cae6e7024	5 years ago
Joshua C. Colp	cce5ee5b7a	res_rtp_asterisk: Force resync on SSRC change. When an SSRC change occurs the timestamps are likely to change as well. As a result we need to reset the timestamp mapping done in the calc_rxstamp function so that they map properly from timestamp to real time. This previously occurred but due to packet retransmission support the explicit setting of the marker bit was not effective. ASTERISK-29352 Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a	5 years ago
Torrey Searle	8c247e2a94	res/res_rtp_asterisk: generate new SSRC on native bridge end For RTCP to work, we update the ssrc to be the one corresponding to the native bridge while active. However when the bridge ends we should generate a new SSRC as the sequence numbers will not continue from the native bridge left off. ASTERISK-29300 #close Change-Id: I23334b6934d2bf6490bda4bbf6414d96b8d17d10	5 years ago
Salah Ahmed	5d42dd2e6a	res_rtp_asterisk: Check remote ICE reset and reset local ice attrb This change will check is the remote ICE session got reset or not by checking the offered ufrag and password with session. If the remote ICE reset session then Asterisk reset its local ufrag and password to reject binding request with Old ufrag and Password. ASTERISK-29266 Change-Id: I9c55e79a7af98a8fbb497d336b828ba41bc34eeb	5 years ago
Kevin Harwell	e5e49d7ecd	res_rtp_asterisk: Add packet subtype during RTCP debug when relevant For some RTCP packet types the report count is actually the packet's subtype. This was not being reflected in the packet debug output. This patch makes it so for some RTCP packet types a "Packet Subtype" is now output in the debug replacing the "Reception reports" (i.e count). Change-Id: Id4f4b77bb37077a4c4f039abd6a069287bfefcb8	5 years ago
Alexander Traud	389b8b0774	rtp: Enable srtp replay protection Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	5 years ago
Sean Bright	e7b13df394	res_rtp_asterisk.c: Fix signed mismatch that leads to overflow ASTERISK-29205 #close Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea	5 years ago
Kevin Harwell	56028426de	Logging: Add debug logging categories Added debug logging categories that allow a user to output debug information based on a specified category. This lets the user limit, and filter debug output to data relevant to a particular context, or topic. For instance the following categories are now available for debug logging purposes: dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet These debug categories can be enable/disable via an Asterisk CLI command. While this overrides, and outputs debug data, core system debugging is not affected by this patch. Statements still output at their appropriate debug level. As well backwards compatibility has been maintained with past debug groups that could be enabled using the CLI (e.g. rtpdebug, stundebug, etc.). ASTERISK-29054 #close Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849	5 years ago
Joshua C. Colp	c84d962eae	res_rtp_asterisk: Don't assume setting retrans props means to enable. The "value" passed in when setting an RTP property determines whether it should be enabled or disabled. The RTP send and receive retrans props did not examine this to know if the buffers should be enabled. They assumed they always should be. This change makes it so that the "value" passed in is respected. ASTERISK-28939 Change-Id: I9244cdbdc5fd065c7f6b02cbfa572bc55c7123dc	5 years ago
sungtae kim	c8c94b6cf1	res_rtp_asterisk.c: Fixed memory leak Added freeifaddrs() for memory releasing. ASTERISK-28904 Change-Id: I109403866e85a30659351946903a679de9727a8f	6 years ago
Guido Falsi	e4366308e1	res_rtp_asterisk: Protect access to nochecksums with #ifdef Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined. ASTERISK-28852 #close Change-Id: I381718893b80599ab8635f2b594a10c1000d595e	6 years ago
Pirmin Walthert	d50fd0acc0	res_rtp_asterisk: Resolve loop when receive buffer is flushed When the receive buffer was flushed by a received packet while it already contained a packet with the same sequence number, Asterisk never left the while loop which tried to order the packets. This change makes it so if the packet is in the receive buffer it is retrieved and freed allowing the buffer to empty. ASTERISK-28827 Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3	6 years ago
Pirmin Walthert	ca032d1e2e	res_rtp_asterisk: Free payload when error on insertion to data buffer When the ast_data_buffer_put rejects to add a packet, for example because the buffer already contains a packet with the same sequence number, the payload will never be freed, resulting in a memory leak. The data buffer will now return an error if this situation occurs allowing the caller to free the payload. The res_rtp_asterisk module has also been updated to do this. ASTERISK-28826 Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1	6 years ago
bernard merindol	7db03e12a7	res_rtp_asterisk.c: Check for first DTMF having timestamp set to 0 When the first DTMF receive in RF2833 codec have TimeStamp at 0 is not processed. ASTERISK-28812 Change-Id: I3196803a062dd2daee4938c9a778c3810cb7e504	6 years ago
Jaco Kroon	2b80e5f5da	res_rtp_asterisk: iterate all local addresses looking to populate ICE. By using pjproject to give us a list of candidates, and then filtering, if the host has >32 addresses configured, then it is possible that we end up filtering out all 32 of those, and ending up with no candidates at all. Instead, get getifaddrs (which pjsip is using underlying anyway) to retrieve all local addresses, and iterate those, adding the first 32 addresses not excluded by the ICE ACL. In our setup at any point in time We've got between 6 and 328 addresses on any given system. The lower limit is the lower limit but the upper limit is growing on a near daily basis currently. Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Alexander Traud	ee1c7f465b	res_rtp_asterisk: Build without PJProject. Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec	6 years ago
traud	1ef1b1b0c2	res_rtp_asterisk: Avoid absolute value on unsigned subtraction. ASTERISK-28809 Change-Id: I269731715347c8e5ef7db1b6ffd3f8d15fc04be4	6 years ago
Joshua C. Colp	96e8d411e1	res_rtp_asterisk: Ensure sufficient space for worst case NACK. ASTERISK-28790 Change-Id: I10df52f98b19ed62575f25dab36e82d136dccd99	6 years ago
Jaco Kroon	82c3939c38	res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses. A pure blacklist is not good enough, we need a whitelist mechanism as well, and the simplest way to do that is to re-use existing ACL infrastructure. This makes it simpler to blacklist say an entire block (/24) except a smaller block (eg, a /29 or even a /32). Normally you'd need to recursively split the block, so if you want to blacklist a /24 except for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I feel that having an ACL instead of a blacklist only is clearer. Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Torrey Searle	a1dba820cf	res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use bridge_p2p_rtp_write will forward rtp to the bridged rtp instance without modifying the ssrc. However, it is not updating the SSRC in the bridged rtp. Thus, when SSRC packets are generated, they have the correct SSRC for the sender. ASTERISK-28773 #close Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478	6 years ago
Joshua Colp	e8468eee13	Merge "res_rtp_asterisk: Add 'rtp show settings' cli command"	6 years ago
Rodrigo Ramírez Norambuena	e089779908	res_rtp_asterisk: Add 'rtp show settings' cli command This change introduce a CLI command for the RTP to display the general configuration. In the first step add the follow fields of the configurations: - rtpstart - rtpend - dtmftimeout - rtpchecksum - strictrtp - learning_min_sequential - icesupport Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f	6 years ago
Joshua C. Colp	87fda066ea	res_rtp_asterisk: Improve video performance in certain networks. The receive buffer will now grow if we end up flushing the receive queue after not receiving the expected packet in time. This is done in hopes that if this is encountered again the extra buffer size will allow more time to pass and any missing packets to be received. The send buffer will now grow if we are asked for packets and can't find them. This is done in hopes that the packets are from the past and have simply been expired. If so then in the future with the extra buffer space the packets should be available. Sequence number cycling has been handled so that the correct sequence number is calculated and used in various places, including for sorting packets and for determining if a packet is old or not. NACK sending is now more aggressive. If a substantial number of missing sequence numbers are added a NACK will be sent immediately. Afterwards once the receive buffer reaches 25% a single NACK is sent. If the buffer continues to grow and reaches 50% or greater a NACK will be sent for each received future packet to aggressively ask the remote endpoint to retransmit. ASTERISK-28764 Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41	6 years ago
Ben Ford	168637cc0c	RTP/ICE: Send on first valid pair. When handling ICE negotiations, it's possible that there can be a delay between STUN binding requests which in turn will cause a delay in ICE completion, preventing media from flowing. It should be possible to send media when there is at least one valid pair, preventing this scenario from occurring. A change was added to PJPROJECT that adds an optional callback (on_valid_pair) that will be called when the first valid pair is found during ICE negotiation. Asterisk uses this to start the DTLS handshake, allowing media to flow. It will only be called once, either on the first valid pair, or when ICE negotiation is complete. ASTERISK-28716 Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867	6 years ago
Kevin Harwell	3865b3fd6a	res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup There was a race condition between client initiated DTLS setup, and handling of server side ice completion that caused the underlying SSL object to get cleared during DTLS initialization. If this happened Asterisk would be left in a partial DTLS setup state. RTP packets were sent and received, but were not being encrypted and decrypted. This resulted in no audio, or static. Specifically, this occurred when '__rtp_recvfrom' was processing the handshake sequence from the client to the server, and then 'ast_rtp_on_ice_complete' gets called from another thread and clears the SSL object when calling the 'dtls_perform_setup' function. The timing had to be just right in the sense that from the external SSL library perspective SSL initialization completed (rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks it is not finished, thus not completing 'dtls_srtp_setup'. This patch removes calls to 'dtls_perform_setup', which clears the SSL object, in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear the underlying SSL object. If an ice candidate changes a full protocol level renegotiation occurs. Also, in the case of bundled ICE candidates are reused when a stream is added. So no real reason to have to clear, and reset in this instance. Also, this patch adds a bit of extra logging to aid in diagnosis of any future problems. ASTERISK-28742 #close Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f	6 years ago

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349 Commits (32b57067a644cb66c2af90704cca5085cc573125)