164 Commits (15)

Author	SHA1	Message	Date
Kevin Harwell	720f76def0	rtp_engine: rtcp_report_to_json can overflow the ssrc integer value ... When writing an RTCP report to json the code attempts to pack the "ssrc" and "source_ssrc" unsigned integer values as a signed int value type. This of course means if the ssrc's unsigned value is greater than that which can fit into a signed integer value it gets converted to a negative number. Subsequently, the negative value goes out in the json report. This patch now packs the value as a json_int_t, which is the widest integer type available on a given system. This should make it so the value no longer overflows. Note, this was caught by two failing tests hep/rtcp-receiver/ and hep/rtcp-sender. Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0	8 years ago
Joshua Colp	4359a386e5	rtp: Don't negotiate dynamic codecs using payload. ... In Asterisk there are some dynamic codecs that have a fixed payload number. This number was being improperly used to negotiate the codec, instead of using the name and sample rate. This could result in the wrong payload number being negotiated for a codec. This change makes it so that only static payloads will be negotiated using their payload number. ASTERISK-27848 Change-Id: Ia865830170fd3f808cdb33104f3d4c4ffdc77570	8 years ago
Joshua Colp	eb52c70f5b	Merge "rtp: Add support for RTP extension negotiation and abs-send-time." into 15	8 years ago
Joshua Colp	d94dd27667	rtp: Add support for RTP extension negotiation and abs-send-time. ... When RTP was originally created it had the ability to place a single extension in an RTP packet. In practice people wanted to potentially put multiple extensions in one and so RFC 5285 (obsoleted by RFC 8285) came into existence. This allows RTP extensions to be negotiated with a unique identifier to be used in the RTP packet, allowing multiple extensions to be present in the packet. This change extends the RTP engine API to add support for this. A user of it can enable extensions and the API provides the ability to retrieve the information (to construct SDP for example) and to provide negotiated information (from SDP). The end result is that the RTP engine can then query to see if the extension has been negotiated and what unique identifier is to be used. It is then up to the RTP engine implementation to construct the packet appropriately. The first extension to use this support is abs-send-time which is defined in the REMB draft[1] and is a second timestamp placed in an RTP packet which is for when the packet has left the sending system. It is used to more accurately determine the available bandwidth. ASTERISK-27831 [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 Change-Id: I508deac557867b1e27fc7339be890c8018171588	8 years ago
Jenkins2	510d9c109c	Merge "rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again." into 15	8 years ago
Alexander Traud	4f60585de1	rtp_engine: Remove the double assigned RTP payload ID of H.263+. ... Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8 assigned another payload ID 98 for this format in Asterisk 1.6. Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667	8 years ago
Alexander Traud	a36d45bdcb	rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. ... This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples of media formats which were affected. ASTERISK-27850 Reported by: Dinis Brazão, Selene Feigl Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496	8 years ago
Joshua Colp	dec428faf3	AST-2018-001: rtp / channel: Don't allow an unnegotiated format to be passed up. ... When an RTP packet is received by an RTP engine it has to map the payload into the Asterisk format. The code was incorrectly checking our own static list for ALL payloads if it couldn't find a negotiated one. This included dynamic payloads. If the payload mapped to a format of a different type (for example receiving a video packet on an audio RTP instance) then the core stream code could cause a crash if a legacy channel driver was in use as no stream would be present. To provide further protection the core stream code will no longer assume that a video or audio frame will always have a stream for legacy channel drivers. If no stream is present the frame is dropped. ASTERISK-27488 Change-Id: I022556f524ad8379ee73f14037040af17ea3316a	9 years ago
Alexander Traud	9b9ee53a06	rtp_engine: Load format name / mime type in uppercase again. ... This reverts a previous change partly. ASTERISK-27689 Change-Id: Ia3d2f282db6995be8c1c253b5d52f6038761e8af	9 years ago
Corey Farrell	e0319bde79	rtp_engine: Add missing unlock. ... Change-Id: I380c31a255e060309f4916da11176e0d00813215	9 years ago
Sean Bright	b64c02e414	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	9 years ago
Richard Mudgett	0074307c61	rtp_engine.c: Eliminate rtcp_report_to_json() RAII_VAR usage. ... Change-Id: I58a22c2ca82e91d7537409b7b3af2d735827a54d	9 years ago
Richard Mudgett	39c8d566ad	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	9 years ago
Corey Farrell	ba3ef46847	RTP Engine: Deal with errors returned from AST_VECTOR_REPLACE. ... Check for errors from AST_VECTOR_REPLACE and clean memory if needed. Change-Id: I124d15cc1d645f85a72a1279f623c1993b304b0b	9 years ago
Sean Bright	ddb8fd6124	dtls: Add support for ephemeral DTLS certificates. ... This mimics the behavior of Chrome and Firefox and creates an ephemeral X.509 certificate for each DTLS session. Currently, the only supported key type is ECDSA because of its faster generation time, but other key types can be added in the future as necessary. ASTERISK-27395 Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4	9 years ago
Sean Bright	1d33757aa0	rtp_engine: Prevent possible double free with DTLS config ... ASTERISK-27225 #close Reported by: Richard Kenner Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89	9 years ago
Joshua Colp	451d86d62e	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. ... This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45	9 years ago
Joshua Colp	927fc6bbd9	core: Add VP9 passthrough support. ... This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc	9 years ago
Joshua Colp	065c3005ad	res_rtp_asterisk / res_pjsip: Add support for BUNDLE. ... BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11	9 years ago
Mark Michelson	d6535c0080	SDP API: Add SSRC-level attributes ... RFC 5576 defines how SSRC-level attributes may be added to SDP media descriptions. In general, this is useful for grouping related SSRCes, indicating SSRC-level format attributes, and resolving collisions in RTP SSRC values. These attributes are used widely by browsers during WebRTC communications, including attributes defined by documents outside of RFC 5576. This commit introduces the addition of SSRC-level attributes into SDPs generated by Asterisk. Since Asterisk does not tend to use multiple SSRCs on a media stream, the initial support is minimal. Asterisk includes an SSRC-level CNAME attribute if configured to do so. This at least gives browsers (and possibly others) the ability to resolve SSRC collisions at offer-answer time. In order to facilitate this, the RTP engine API has been enhanced to be able to retrieve the SSRC and CNAME on a given RTP instance. res_rtp_asterisk currently does not provide meaningful CNAME values in its RTCP SDES items, and therefore it currently will always return an empty string as the CNAME value. A task in the near future will result in res_rtp_asterisk generating more meaningful CNAMEs. Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789	9 years ago
Jenkins2	e478d2eb94	Merge "res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP."	9 years ago
Richard Mudgett	d165079cbc	rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes. ... The struct ast_rtp_instance has historically been indirectly protected from reentrancy issues by the channel lock because early channel drivers held the lock for really long times. Holding the channel lock for such a long time has caused many deadlock problems in the past. Along comes chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock because sometimes there may not be an associated channel created yet or the channel pointer isn't available. In the case of ASTERISK-26835 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up changing the rtp->rtcp->local_addr_str string and interfering with each other. The classic reentrancy problem resulted in a crash. In the case of ASTERISK-26853 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up processing ICE candidates in PJPROJECT and interfering with each other. The classic reentrancy problem resulted in a crash. * rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP instance struct. * rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP instance struct for the API call. * res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy problem with rtp->rtcp->local_addr_str in the scheduler thread running ast_rtcp_write(). * res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in bridge_p2p_rtp_write() because there are two RTP instance structs involved. * res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler callbacks. We cannot hold the instance lock when trying to stop a scheduler callback. * res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads to prevent a race condition between starting and stopping a timeout timer. The race condition is no longer present between dtls_perform_handshake() and __rtp_recvfrom() because the instance lock prevents these functions from overlapping each other with regards to the timeout timer. * res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads using a condition signal to know when TURN negotiations complete. * res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN ioqueue_worker_thread(). We cannot hold the instance lock when trying to create or shut down the worker thread without a risk of deadlock. This patch exposed a race condition between a PJSIP serializer thread setting up an ICE session in ice_create() and another thread reading RTP packets. * res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we have re-locked the RTP instance to prevent the other thread from trying to process ICE packets on an incomplete ICE session setup. A similar race condition is between a PJSIP serializer thread resetting up an ICE session in ice_create() and the timer_worker_thread() processing the completion of the previous ICE session. * res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an uninitialized/null remote_address after calling update_address_with_ice_candidate(). * res_rtp_asterisk.c: Eliminate the chance of ice_reset_session() destroying and setting the rtp->ice pointer to NULL while other threads are using it by adding an ao2 wrapper around the PJPROJECT ice pointer. Now when we have to unlock the RTP instance object to call a PJPROJECT ICE function we will hold a ref to the wrapper. Also added some rtp->ice NULL checks after we relock the RTP instance and have to do something with the ICE structure. ASTERISK-26835 #close ASTERISK-26853 #close Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4	9 years ago
Alexander Traud	72c5f3b0ba	res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP. ... This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compact_headers=yes via the file pjsip.conf. ASTERISK-26932 #close Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689	9 years ago
Kevin Harwell	9b103e7bea	rtp_engine: allocate RTP dynamic payloads per session ... Dynamic payload types were statically defined in Asterisk. This unfortunately limited the number of dynamic payloads that could be registered. With this patch dynamic payload type numbers are now assigned dynamically and per RTP instance. However, in order to limit any issues where some clients expect the old statically defined value this patch makes it so the value Asterisk used to pre- designate is used for the dynamic assignment if available. An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf) that turns the new dynamic behavior on or off. When off it reverts back to using statically defined payload values. This option defaults to "yes" in Asterisk 15. ASTERISK-26515 #close patches: ASTERISK-26515.diff submitted by jcolp (license 5000 Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc	10 years ago
George Joseph	8470c2bdea	RFC sdp: Initial SDP creation ... * Added additional fields to ast_sdp_options. * Re-organized ast_sdp. * Updated field names to correspond to RFC4566 terminology. * Created allocs/frees for SDP children. * Created getters/setters for SDP children where appropriate. * Added ast_sdp_create_from_state. * Refactored res_sdp_translator_pjmedia for changes. Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48	10 years ago
Richard Mudgett	16fdb11bc3	core: Cleanup some channel snapshot staging anomalies. ... We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	10 years ago
gestoip2	d9b24cce0a	res_rtp_asterisk: RTT miscalculation in RTCP ... When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. RTT calculation is correct, but the data representation isn't. RTT is represented by a 32-bit fixed-point number with the integer part in the first 16 bits and the fractional part in the last 16 bits. In order to get the RTT value, the fractional part is miscalculated, there is an unnecessary 16 bit shift that causes overflow. Besides this there is another mistake, when transforming the integer value to the fixed point fractional part via bitwise operation, that loses precision. * RTT fractional part is no longer shifted, avoiding overflow. * RTT fractional part is transformed to its fixed-point value more precisely. * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. * Fixed NTP timestamp report logging. The usec was inexplicably multiplied by 4096. ASTERISK-26566 #close Reported by Hector Royo Concepcion Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f	10 years ago
Alexander Traud	9ac53877f6	rtp_engine: Allow more than 32 dynamic payload types. ... Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK (Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, this change utilizes payload types in the range between 35 and 63 giving room for another 29 payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964	10 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	10 years ago
Joshua Colp	0376af9519	rtp: Only accept the first payload for a format in SDP. ... When receiving an SDP offer with multiple payloads for the same format we would generate an answer with the first payload, but during the payload crossover operation (to set the payloads for receiving) we would remove all payloads but the last. This would result in incoming traffic being matched against the wrong format and outgoing traffic being sent using the wrong payload. This change makes it so that once a format has a payload number put into the mapping all subsequent ones are ignored. This ensures there is only ever one payload in the mapping and that it is the payload placed into the answer SDP. ASTERISK-26365 #close Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60	10 years ago
Matt Jordan	a19f4affe8	manager: Add <see-also> links between related events ... This patch adds some see-also references between related AMI events. It focuses primarily on those events that are guaranteed to come in pairs, such as DTMFBegin/DTMFEnd, as well as those that occur during the life cycle of an Asterisk channel, such as Newchannel/Hangup. Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3	10 years ago
Mark Michelson	1cd79d6ee5	Remove SILK payload mappings from Asterisk core. ... SILK is a bit of a hog when it comes to using up our limited number of dynamic payload types in the RTP engine. By freeing up four slots, it allows for other codecs to potentially take the place. Now, codec_silk.so will dynamically use the payload slots in the RTP engine when it loads. A better fix would be make RTP dynamic payload types actually dynamic. However, at this stage of Asterisk 14 development, this is a risky move that would be imprudent. Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612	10 years ago
Kevin Harwell	1d364ac54f	rtp_engine: Failed assertion and wrong name given for codec ... Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c	10 years ago
Mark Michelson	273052f404	Update support for SILK format. ... This commit adds scaffolding in order to support the SILK audio format on calls. Roughly, this is what is added: * Cached silk formats. One for each possible sample rate. * ast_codec structures for each possible sample rate. * RTP payload mappings for "SILK". In addition, this change overhauls the res_format_attr_silk file in the following ways: * The "samplerate" attribute is scrapped. That's native to the format. * There are far more checks to ensure that attributes have been allocated before attempting to reference them. * We do not SDP fmtp lines for attributes set to 0. These changes make way to be able to install a codec_silk module and have it actually work. It also should allow for passthrough silk calls in Asterisk. Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e	10 years ago
Jacek Konieczny	970803efcb	res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS ... Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764 explicitly states: There MUST be a separate DTLS-SRTP session for each distinct pair of source and destination ports used by a media session This means RTP keying material cannot be used for DTLS RTCP, which was the reason why RTCP encryption would fail. ASTERISK-25642 Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a	11 years ago
Joshua Colp	38bf1f3a71	Merge "json: Audit ast_json_* usage for thread safety."	11 years ago
Joshua Colp	a2431f83ef	rtp_engine: Ignore empty filenames in DTLS configuration. ... When applying an empty DTLS configuration the filenames in the configuration will be empty. This is actually valid to do and each filename should simply be ignored. Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539	11 years ago
Joshua Colp	d17d9a9288	json: Audit ast_json_* usage for thread safety. ... The JSON library Asterisk uses, jansson, is not thread safe for us in a few ways. To help with this wrappers for JSON object reference count increasing and decreasing were added which use a global lock to ensure they don't clobber over each other. This does not extend to reference count manipulation within the jansson library itself. This means you can't safely use the object borrowing specifier (O) in ast_json_pack and you can't share JSON instances between objects. This change removes uses of the O specifier and replaces them with the o specifier and an explicit ast_json_ref. Some cases of instance sharing have also been removed. ASTERISK-25601 #close Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1	11 years ago
George Joseph	a987434564	res_pjsip: Add existence and readablity checks for tls related files ... Both transport and endpoint now check for the existence and readability of tls certificate and key files before passing them on to pjproject. This will cause the object to not load rather than waiting for pjproject to discover that there's a problem when a session is attempted. NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located in build_peer which is gigantic and I didn't want to disturb it. Error messages will emit but it won't interrupt chan_sip loading. ASTERISK-25618 #close Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 Reported-by: George Joseph Tested-by: George Joseph	11 years ago
Alexander Traud	525c7ab780	rtp_engine: Init a format-attribute module to its RFC defaults. ... Previously, format-attribute modules relied on an existing fmtp line in SDP negotiation. However, fmtp is optional for several formats like the Opus Codec. Now, the format-attribute module is called with an empty fmtp, which allows the module to initialise itself to RFC defaults. Furthermore now, Asterisk is able to differentiate between internally and externally created formats. ASTERISK-25537 #close Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52	11 years ago
Richard Mudgett	f7df3e1a01	rtp_engine.c: Get current or create a needed rx payload type mapping. ... * Make ast_rtp_codecs_payload_code() get the current mapping or create a rx payload type mapping. ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: Ia4b2d45877a8f004f6ce3840e3d8afe533384e56	11 years ago
Richard Mudgett	38854a9f7b	rtp_engine.c: Extract rtp_codecs_payload_replace_rx(). ... ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: I34e23bf5b084c8570f9c3e6ccd19b95fe85af239	11 years ago
Richard Mudgett	1a549ed134	rtp_engine.c: Initial split of payload types into rx and tx mappings. ... There are numerous problems with the current implementation of the RTP payload type mapping in Asterisk. It uses only one mapping structure to associate payload types to codecs. The single mapping is overkill if all of the payload type values are well known values. Dynamic payload type mappings do not work as well with the single mapping because RFC3264 allows each side of the link to negotiate different dynamic mappings for what they want to receive. Not only could you have the same codec mapped for sending and receiving on different payload types you could wind up with the same payload type mapped to different codecs for each direction. 1) An independent payload type mapping is needed for sending and receiving. 2) The receive mapping needs to keep track of previous mappings because of the slack to when negotiation happens and current packets in flight using the old mapping arrive. 3) The transmit mapping only needs to keep track of the current negotiated values since we are sending the packets and know when the switchover takes place. * Needed to create ast_rtp_codecs_payload_code_tx() and make some callers use the new function because ast_rtp_codecs_payload_code() was used for mappings in both directions. * Needed to create ast_rtp_codecs_payloads_xover() for cases where we need to pass preferred codec mappings to the peer channel for early media bridging or when we need to prefer the offered mapping that RFC3264 says we SHOULD use. * ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are the only new public functions created. All the others were only used for the tx or rx mapping direction so the function doxygen now reflects which direction the function operates. * chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing that makes no sense when processing an incoming SDP. We would be wiping out any mappings that we set for the possible outgoing SDP we sent earlier. ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac	11 years ago
Richard Mudgett	f49bef08a2	rtp_engine.c: Fix performance issue with several channel drivers that use RTP. ... ast_rtp_codecs_get_payload() gets called once or twice for every received RTP frame so it would be nice to not allocate an ao2 object to then have it destroyed shortly thereafter. The ao2 object gets allocated only if the payload type is not set by the channel driver as a negotiated value. The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323. * Made static_RTP_PT[] an array of ao2 objects that ast_rtp_codecs_get_payload() can return instead of an array of structs that must be copied into a created ao2 object. ASTERISK-25296 #close Reported by: Richard Mudgett Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0	11 years ago
Richard Mudgett	5f925d48b7	rtp_engine.c: Must protect mime_types_len with mime_types_lock. ... Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e	11 years ago
Richard Mudgett	e2d5d4db35	rtp_engine.h: No sense allowing payload types larger than RFC allows. ... * Tweaked add_static_payload() to not use magic numbers. Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b	11 years ago
Richard Mudgett	bc1eae55cb	rtp_engine.c: Minor tweaks. ... * Fix off nominial ref leak of new_type in ast_rtp_codecs_payloads_set_m_type(). * No need to lock static_RTP_PT_lock in ast_rtp_codecs_payloads_set_m_type() and ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type parameter sanity check. * No need to create ast_rtp_payload_type ao2 objects with a lock since the lock is not used. Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4	11 years ago
Joshua Colp	309dd2a409	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	11 years ago
Mark Michelson	2b42264e66	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d	11 years ago
Walter Doekes	13a318bbb1	rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format. ... When running valgrind on Asterisk, it complained about: ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304) ==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...) ==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292) ==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437) The code in question is a struct assignment, which may be performed by memcpy as a compiler optimization. It is changed to only copy the struct contents if source and destination are different. ASTERISK-25219 #close Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a	11 years ago