245 Commits (b2987774588098248830917819586039fcb96882)

Author	SHA1	Message	Date
Torrey Searle	7c17bc75ed	res/res_rtp_asterisk: clear smoother when local bridging ... p2p_write updates txformat but doesn't require a smoother. If a smoother was created by another bridge type the smoother could fall out of date causing one way audio issues. To prevent this the smoother is now destroyed on the start of native bridge. ASTERISK-28284 #close Change-Id: I84e67f144963787fff9b4d79ac500514fb40cdc6	7 years ago
Alexei Gradinari	f9face3612	RTP: reset DTMF last seqno/timestamp on RTP renegotiation ... The remote side may start a new stream when renegotiating RTP. Need to reset the DTMF last sequence number and the timestamp of the last END packet on RTP renegotiation. If the new time stamp is lower then the timestamp of the last DTMF END packet the asterisk drops all DTMF frames as out of order. This bug was caught using Cisco ip-phone SPA5XX and codec g722. On SIP session update the SPA50X resets stream and a new timestamp is twice smaller then the previous. ASTERISK-28162 #close Change-Id: Ic72b4497e74d801b27a635559c1cf29c16c95254	7 years ago
Sean Bright	a2336adfb9	res_rtp_asterisk: Remove some unused structure fields. ... All of the fields that were removed were no longer referenced except for 'lastrxts' and 'rxseqno' which were only ever written to. Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c	7 years ago
Friendly Automation	212f2a3aa3	Merge "Use non-blocking socket() and pipe() wrappers" into 13	7 years ago
Friendly Automation	53bda83f5e	Merge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"" into 13	7 years ago
Sean Bright	b84fd37ef0	Use non-blocking socket() and pipe() wrappers ... Change-Id: I050ceffe5a133d5add2dab46687209813d58f597	7 years ago
George Joseph	1ef993911f	Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit" ... This reverts commit cf620ce0f6. Pending resolution of ASTERISK_28200 Change-Id: If18ab1166db0d634ce8a099bd7460251e9c682e1	7 years ago
Jenkins2	9f0459d2a0	Merge "RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit" into 13	7 years ago
Corey Farrell	0a9904e1c6	astobj2: Eliminate usage of legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. ao2_container_alloc is now restricted to modules only and is being removed from Asterisk 17. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Alexei Gradinari	cf620ce0f6	RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit ... The marker bit set on the voice packet indicates the start of a new stream and a new time stamp. Need to reset the DTMF last sequence number and the timestamp of the last END packet. If the new time stamp is lower then the timestamp of the last DTMF END packet the asterisk drops all DTMF frames as out of order. This bug was caught using Cisco ip-phone SPA50X and codec g722. On SIP session update the SPA50X resets stream indicating it with market bit and a new timestamp is twice smaller then the previous. ASTERISK-28162 #close Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620	7 years ago
Ben Ford	aa31657e28	res_rtp_asterisk.c: Add "seqno" strictrtp option ... When networks experience disruptions, there can be large gaps of time between receiving packets. When strictrtp is enabled, this created issues where a flood of packets could come in and be seen as an attack. Another option - seqno - has been added to the strictrtp option that ignores the time interval and goes strictly by sequence number for validity. Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71	7 years ago
George Joseph	efadf89131	Merge "res_rtp_asterisk: Raise event when RTP port is allocated" into 13	7 years ago
Joshua Colp	155ff8e174	res_rtp_asterisk: Raise event when RTP port is allocated ... This change raises a testsuite event to provide what port Asterisk has actually allocated for RTP. This ensures that testsuite tests can remove any assumption of ports and instead use the actual port in use. ASTERISK-28070 Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044	7 years ago
Sean Bright	3033242a7b	res_rtp_asterisk: Reset all settings on module reload ... 'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults if they are not present in the updated configuration file. Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670	7 years ago
neutrino88	49d388bfa8	res/res_rtp_asterisk: remove debug traces generated by an empty frame ... The realtime text timer pops regularly and sends text frames even if the buffer is empty. This causes a lot of unecessary debug logging. * Made red_write() test if we need to send a frame before calling ast_rtp_write() ASTERISK-28002 Reported by: Emmanuel BUU Tested by: Emmanuel BUU Change-Id: Icf81310c3b8080b615a42060afc02ab41f9523dd	7 years ago
Alexander Traud	a4b105c99b	res_rtp_asterisk: In Developer Mode, do not require OpenSSL. ... OpenSSL is an optional external library and should stay optional even when Developer Mode is configured. ASTERISK-27990 Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b	7 years ago
neutrino88	e5721b01f6	res_rtp_asterisk: Avoid merging command and regular T.140 text packets ... When realtime text packets are to be sent, the text is accumulated in a buffer and sent regularly by a timer. It can happen that commands such as a backspace, CR, or LF get merged with regular text. This breaks some UAs. The proposed change: * We test if the current packet contains a command. If so we send the buffer immediately. * We test if the buffer contained a command. If so we send the buffer immediately. * We accumulate the text (or the command) in the buffer. ASTERISK-27970 Change-Id: Ifbe993311410fa855cb8aa4a12084db75f413462	7 years ago
Alexander Traud	e3de4bc46e	res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP. ... Previously, Asterisk used its script ./configure, to test whether OpenSSL was built with no-srtp (or was simply too old). However, the header file <openssl/opensslconf.h> is the preferred way to detect the local configuration of OpenSSL. As a positive side-effect the script ./configure does not interleave the detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of individual features of OpenSSL anymore. Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0	7 years ago
Alexander Traud	ef2386fcd7	res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. ... Furthermore, allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537	8 years ago
Torrey Searle	4b7fd478d5	res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change ... Certain race conditions between changing bridge types and DTMF can cause the current FLAG_NEED_MARKER_BIT to send the marker bit before the actual first packet of native bridging. This logic keeps track of the ssrc the bridge is currently sending and will correctly ensure the marker bit is set if SSRC as changed from the previous sent packet. ASTERISK-27845 Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b	8 years ago
Corey Farrell	b81eadcefc	Replace direct checks of option_debug with DEBUG_ATLEAST macro. ... Checking option_debug directly is incorrect as it ignores file/module specific debug settings. This system-wide change replaces nearly all direct checks for option_debug with the DEBUG_ATLEAST macro. Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0	8 years ago
Richard Mudgett	104468ad3a	pjproject: Add cache_pools debugging option. ... The pool cache gets in the way of finding use after free errors of memory pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a pool is released because it gets put into the cache instead of being freed. * Added the "cache_pools" option to pjproject.conf. Disabling the option helps track down pool content mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the pool contents are used after free and who freed it. To disable the pool caching simply disable the cache_pools option in pjproject.conf and restart Asterisk. Sample pjproject.conf setting: [startup] cache_pools=no * Made current users of the caching pool factory initialization and destruction calls call common routines to create and destroy cached pools. ASTERISK-27704 Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828	8 years ago
Thomas Guebels	9325de11f8	res_rtp_asterisk: Fix ICE candidate nomination ... If the ICE role is not set right away, we might have a role conflict that stays undetected and ICE finishing with successful tests and no candidate nominated. This was introduced by ASTERISK-27088. To avoid this, we set the role as soon as before but only if the ICE state permits it: still checking and not yet nominating candidates or completed. ASTERISK-27646 Change-Id: I5dbc69ad63cacbb067922850fbb113d479bd729c	8 years ago
Aaron An	c38b750810	res_rtp_asterisk: Avoid close the rtp/rtcp fd twice. ... When RTCP-MUX enabled. rtp->s is the same as rtcp->s, check this before close the file descriptor. Close the FD twice will hangs the asterisk under heavy load. ASTERISK-27299 #close Reported-by: Aaron An Tested-by: AaronAn Change-Id: I870a072d73fd207463ac116ef97100addbc0820a	8 years ago
Richard Mudgett	61e81338d9	res_rtp_asterisk.c: Disable packet flood detection for video streams. ... We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70	8 years ago
Joshua Colp	5705e8ae0e	AST-2017-012: Place single RTCP report block at beginning of report. ... When the RTCP code was transitioned over to Stasis a code change was made to keep track of how many reports are present. This count controlled where report blocks were placed in the RTCP report. If a compound RTCP packet was received this logic would incorrectly place a report block in the wrong location resulting in a write to an invalid location. This change removes this counting logic and always places the report block at the first position. If in the future multiple reports are supported the logic can be extended but for now keeping a count serves no purpose. ASTERISK-27382 ASTERISK-27429 Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116	8 years ago
Sean Bright	f726f11974	utils: Add convenience function for setting fd flags ... There are many places in the code base where we ignore the return value of fcntl() when getting/setting file descriptior flags. This patch introduces a convenience function that allows setting or clearing file descriptor flags and will also log an error on failure for later analysis. Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7	8 years ago
Richard Mudgett	9330eacc50	res_rtp_asterisk.c: Increase strictrtp learning timeout time. ... More complicated direct media reinvite negotiations can result in longer delays before direct media flows. The strictrtp learning timeout time was too short. One log showed that the first RTP packet came in just after three seconds. * Increase the strictrtp learning timeout time from 1.5 to 5 seconds. ASTERISK-27453 Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c	8 years ago
Jenkins2	2edbaa1588	Merge "res_rtp_asterisk.c: Fix rtp source address learning for broken clients" into 13	8 years ago
Alexander Traud	6ebe03d4b3	res_rtp_asterisk: ICE server-reflexive candidates (srflx) with Dual-Stack. ... Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now, srflx is gathered and sent via SDP, even when Asterisk is enabled for Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used. ASTERISK-27437 Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1	8 years ago
Pirmin Walthert	6b0963c42a	res_rtp_asterisk.c: Fix rtp source address learning for broken clients ... Some clients do not send rtp packets every ptime ms. This can lead to situations in which the rtp source learning algorithm will never learn the address of the client. This has been discovered on a Mac mini with a pjsip based softphone after updating to Sierra: as soon as USB headsets are involved, the softphone will send the second packet 30ms after the first, the third 30ms after the second and the fourth 1ms after the third. So in the old implmentation the rtp source learning algorithm was repeatedly reset on the fourth packet. The patch changes the algorithm in a way that doesn't take the arrival time between two consecutive packets into account but the time between the first and the last packet of a learning sequence. The patch also fixes a second problem: when a user was using a wrong value for the probation setting there was a LOG_WARNING output stating that the value had been set to the default value instead. However the code for setting the value back to defaults was missing. ASTERISK-27421 #close Change-Id: If778fe07678a6fd2041eaca7cd78267d0ef4fc6c	8 years ago
Tzafrir Cohen	944e500e95	declare optional openssl dependencies in moduleinfo ... Declare optional openssl dependencies in: * res_rtp_asterisk.c * tcptls.c ASTERISK-27328 #close Change-Id: I2636f1c05b8104b4fe6f36cce0ebd9a98b9c78ab	8 years ago
Richard Mudgett	221d8a5c24	res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential. ... The bridge_p2p_rtp_write() has potential reentrancy problems. * Accessing the bridged RTP members must be done with the instance1 lock held. The DTMF and asymmetric codec checks must be split to be done with the correct RTP instance struct locked. i.e., They must be done when working on the appropriate side of the point to point bridge. * Forcing the RTP mark bit was referencing the wrong side of the point to point bridge. The set mark bit is used everywhere else to set the mark bit when sending not receiving. The patches for ASTERISK_26745 and ASTERISK_27158 did not take into account that not everything carried by RTP uses a codec. The telephony DTMF events are not exchanged with a codec. As a result when RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is enabled, the DTMF digits would always get passed to the core even though the local native RTP bridge is active, and the DTMF digits would go out using the wrong SSRC id. * Add protection for non-format payload types like DTMF when updating the lastrxformat and lasttxformat. Also protect against non-format payload types when checking for asymmetric codecs. ASTERISK-27292 Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186	8 years ago
Richard Mudgett	6d4b801c83	AST-2017-008: Improve RTP and RTCP packet processing. ... Validate RTCP packets before processing them. * Validate that the received packet is of a minimum length and apply the RFC3550 RTCP packet validation checks. * Fixed potentially reading garbage beyond the received RTCP record data. * Fixed rtp->themssrc only being set once when the remote could change the SSRC. We would effectively stop handling the RTCP statistic records. * Fixed rtp->themssrc to not treat a zero value as special by adding rtp->themssrc_valid to indicate if rtp->themssrc is available. ASTERISK-27274 Make strict RTP learning more flexible. Direct media can cause strict RTP to attempt to learn a remote address again before it has had a chance to learn the remote address the first time. Because of the rapid relearn requests, strict RTP could latch onto the first remote address and fail to latch onto the direct media remote address. As a result, you have one way audio until the call is placed on and off hold. The new algorithm learns remote addresses for a set time (1.5 seconds) before locking the remote address. In addition, we must see a configured number of remote packets from the same address in a row before switching. * Fixed strict RTP learning from always accepting the first new address packet as the new stream. * Fixed strict RTP to initialize the expected sequence number with the last received sequence number instead of the last transmitted sequence number. * Fixed the predicted next sequence number calculation in rtp_learning_rtp_seq_update() to handle overflow. ASTERISK-27252 Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c	8 years ago
Richard Mudgett	044674c0cd	res_rtp_asterisk.c: Add doxygen to RTCP payload types. ... Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998	8 years ago
Richard Mudgett	6c922b3157	res_rtp_asterisk.c: Check RTP packet version earlier. ... Change-Id: Ic6493a7d79683f3e5845dff1cee49445fd5a0adf	8 years ago
Joshua Colp	1a022285dd	res_rtp_asterisk: Only learn a new source in learn state. ... This change moves the logic which learns a new source address for RTP so it only occurs in the learning state. The learning state is entered on initial allocation of RTP or if we are told that the remote address for the media has changed. While in the learning state if we continue to receive media from the original source we restart the learning process. It is only once we receive a sufficient number of RTP packets from the new source that we will switch to it. Once this is done the closed state is entered where all packets that do not originate from the expected source are dropped. The learning process has also been improved to take into account the time between received packets so a flood of them while in the learning state does not cause media to be switched. Finally RTCP now drops packets which are not for the learned SSRC if strict RTP is enabled. ASTERISK-27013 Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c	8 years ago
Jenkins2	9bc114ffba	Merge "res_rtp_asterisk: enable rtcp & QOS stats on native bridge" into 13	8 years ago
Torrey Searle	3608f96ea3	res_rtp_asterisk: enable rtcp & QOS stats on native bridge ... Asterisk wasn't generating or forwarding RTCP packets when native bridge was activated. Also the stats weren't available via CHANNEL(qos). Now the RTCP stats are always calculated. ASTERISK-27158 #close Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b	8 years ago
Torrey Searle	c4f201cd73	res_rtp_asterisk: Make P2P bridge Asymmetric codec aware ... Introduce a new property to rtp-engine to make it aware of the desire for assymetric codecs or not. If asymmetric codecs is not allowed, the bridge will compare read/write formats and shut down the p2p bridge if needed ASTERISK-26745 #close Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f	8 years ago
Sean Bright	c16000f201	res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours ... Change-Id: Ia578ede1a55b21014581793992a429441903278b	8 years ago
Joshua Colp	51761b759d	res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use. ... This change makes it so that if an RTCP packet is being sent the RTP ICE component is used for sending if RTCP-MUX is in use. ASTERISK-27133 Change-Id: I6200f611ede709602ee9b89501720c29545ed68b	8 years ago
Richard Mudgett	aa514f420b	res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock. ... When a message is received on the TURN socket, the code processing the message needs to call into the ICE/STUN session for further processing. This code path locks the TURN group lock then the ICE/STUN group lock. In another thread an ICE/STUN timer can fire off to send a keep alive message over the TURN socket. In this code path, the ICE/STUN group lock is obtained then the TURN group lock is obtained to send the packet. A classic deadlock case if the group locks are not the same. * Made TURN get created using the ICE/STUN session's group lock. NOTE: I was originally concerned that the ICE/STUN session can get recreated by ice_reset_session() for an event like RTCP multiplexing causing a change during SDP negotiation. In this case the TURN group lock would become different. However, TURN is also recreated as part of the ICE/STUN recreation in ice_create() when all known ICE candidates are added to the new ICE session. While the ICE/STUN and TURN sessions are being recreated there is a period where the group locks could be different. ASTERISK-27023 #close Patches: res_rtp_asterisk-turn-deadlock-fix.patch (license #6502) patch uploaded by Michael Walton (modified) Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9	8 years ago
Joshua Colp	0426b1d88a	res_rtp_asterisk: Fix issues with ICE renegotiation. ... When re-inviting to add more streams it is possible for the role of existing ICE sessions to be changed to the incorrect value. This results in subsequent refreshes within the sessions getting a role conflict and the ICE session breaking down. This change only sets the role to be the new value if an ICE renegotiation is actually going to happen, otherwise the existing role is preserved. As well if we encounter a situation where a unidirectional ICE negotiation happens and the other side does not send us candidates we will not store any information for sending traffic, even though we know where they are reachable. This change fixes this by using the source of the ICE traffic itself as the target if no candidates are known and we receive some ICE traffic. ASTERISK-27088 Change-Id: I71228181e358917fcefc3100fad21b2fc02a59a9	8 years ago
George Joseph	7dafe82751	res_rtp_asterisk: Fix ssrc change for rtcp srtp ... It looks like there was a copy/paste error in ast_rtp_change_source where if there was a rtcp srtp instance, instead of updating its ssrc we were updating the srtp instance ssrc twice. ASTERISK-27022 #close Reported-by: Michael Walton Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095	8 years ago
Sean Bright	70e5887906	format: Reintroduce smoother flags ... In review 4843 (ASTERISK-24858), we added a hack that forced a smoother creation when sending signed linear so that the byte order was adjusted during transmission. This was needed because smoother flags were lost during the new format work that was done in Asterisk 13. Rather than rolling that same hack into res_rtp_multicast, re-introduce smoother flags so that formats can dictate their own options. Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16	9 years ago
Kevin Harwell	e91efef2bb	res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm ... When using rtcp mux if an rtcp payload came in it would still use the srtp unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp data was being passed to the rtp unprotect method this would result in an error. This patch ensures that the correct unprotect method is chosen by making sure the passed in rtcp flag is appropriately set when rtcp mux is enabled and an rtcp payload is received. ASTERISK-26979 #close Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241	9 years ago
Joshua Colp	ce1985b099	Merge "res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures" into 13	9 years ago
Kevin Harwell	bbe90d6aed	res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures ... When a call gets put on hold RTP is temporarily stopped and Asterisk was setting the remote RTCP address to NULL. Then when RTCP data was received from the remote endpoint, Asterisk would be missing this information when publishing the rtcp_message stasis event. Consequently, message subscribers (in this case res_hep_rtcp) trying to parse the "from" field output the following error: "ast_sockaddr_split_hostport: Port missing in (null)" This patch makes it so the remote RTCP address is no longer set to NULL when stopping RTP. There was only one place that appeared to check if the remote RTCP address was NULL as a way to tell if RTCP was running. This patch added an additional check on the RTCP schedid for that case to make sure RTCP was truly not running. ASTERISK-26860 #close Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b	9 years ago
Richard Mudgett	55f452884f	res_rtp_asterisk.c: Fix crash in RTCP DTLS operation. ... Occasionally a crash happens when processing the RTCP DTLS timeout handler. The RTCP DTLS timeout timer could be left running if we have not completed the DTLS handshake before we place the call on hold or we attempt direct media. * Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP. * Made some sanity tweaks to ast_rtp_prop_set() when switching from standard RTCP mode to RTCP multiplexed mode. ASTERISK-26692 #close Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0	9 years ago