150 Commits (6fb811a5905a54a07a901b25f8f3af0ad9d3f7c9)

Author	SHA1	Message	Date
George Joseph	02e2000653	res_pjsip: Add new endpoint option "suppress_moh_on_sendonly" ... Normally, when one party in a call sends Asterisk an SDP with a "sendonly" or "inactive" attribute it means "hold" and causes Asterisk to start playing MOH back to the other party. This can be problematic if it happens at certain times, such as in a 183 Progress message, because the MOH will replace any early media you may be playing to the calling party. If you set this option to "yes" on an endpoint and the endpoint receives an SDP with "sendonly" or "inactive", Asterisk will NOT play MOH back to the other party. Resolves: #979 UserNote: The new "suppress_moh_on_sendonly" endpoint option can be used to prevent playing MOH back to a caller if the remote end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.	11 months ago
Mike Bradeen	055031dfcb	res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch ... When Asterisk sends an offer to Bob that includes 48K and 8K codecs with matching 4733 offers, Bob may want to use the 48K audio codec but can not accept 48K digits and so negotiates for a mixed set. Asterisk will now check Bob's offer to make sure Bob has indicated this is acceptible and if not, will use Bob's preference. Fixes: #847	1 year ago
Alexei Gradinari	2acd9982de	res_pjsip_sdp_rtp fix leaking astobj2 ast_format ... PR #700 added a preferred_format for the struct ast_rtp_codecs, but when set the preferred_format it leaks an astobj2 ast_format. In the next code ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0)); both functions ast_rtp_codecs_set_preferred_format and ast_format_cap_get_format increases the ao2 reference count. Fixes: #856	1 year ago
Tinet-mucw	291c1e4c1f	res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto ... When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling. Resolves: asterisk#826	1 year ago
Mike Bradeen	52bdfe09ab	res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits ... After change made in 624f509 to add support for non 8K RFC 4733/2833 digits, Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of the negotiated codec(s). This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec. A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to be re-written to allow for these scenarios. Fixes: #776	1 year ago
Mike Bradeen	3b4d2a6506	rtp_engine: add support for multirate RFC2833 digits ... Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs. Asterisk currently treats RFC2833 Digits as a single rtp payload type with a fixed bitrate of 8K. This change would expand that to 8, 16, 24 and 32K. This requires checking the offered rtp types for any of these bitrates and then adding an offer for each (if configured for RFC2833.) DTMF generation must also be changed in order to look at the current outbound codec in order to generate appropriately timed rtp. For cases where no outgoing audio has yet been sent prior to digit generation, Asterisk now has a concept of a 'preferred' codec based on offer order. On inbound calls Asterisk will mimic the payload types of the RFC2833 digits. On outbound calls Asterisk will choose the next free payload types starting with 101. UserNote: No change in configuration is required in order to enable this feature. Endpoints configured to use RFC2833 will automatically have this enabled. If the endpoint does not support this, it should not include it in the SDP offer/response. Resolves: #699	1 year ago
Henrik Liljedahl	b267629b77	res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeout setting. ... First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation. Fixes #710	1 year ago
Ben Ford	1adefb886a	res_pjsip_sdp_rtp.c: Use correct timeout when put on hold. ... When a call is put on hold and it has moh_passthrough and rtp_timeout set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is expected to be used, but rtp_timeout is used instead. This change adds a couple of checks for locally_held to determine if rtp_timeout_hold needs to be used instead of rtp_timeout. ASTERISK-30350 Change-Id: I7b106fc244332014216d12bba851cefe884cc25f	3 years ago
Maximilian Fridrich	14826a8038	res_pjsip: Add mediasec capabilities. ... This patch adds support for mediasec SIP headers and SDP attributes. These are defined in RFC 3329, 3GPP TS 24.229 and draft-dawes-sipcore-mediasec-parameter. The new features are implemented so that a backbone for RFC 3329 is present to streamline future work on RFC 3329. With this patch, Asterisk can communicate with Deutsche Telekom trunks which require these fields. ASTERISK-30032 Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2	3 years ago
Joshua C. Colp	33d38a8d6f	res_pjsip_sdp_rtp: Skip formats without SDP details. ... When producing an outgoing SDP we iterate through the configured formats and produce SDP information. It is possible for some configured formats to not have SDP information available. If this is the case we skip over them to allow the SDP to still be produced. ASTERISK-29185 Change-Id: I3e37569aa4ca341260e6ca5904dc2f75e46a1749	3 years ago
Boris P. Korzun	6624e34580	res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity ... Change RTP timer behavior for detecting RTP only after two-way SDP channel establishment. Ignore detecting after receiving 183 with SDP or while direct media is used. Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout and rtpholdtimeout options in chan_sip. ASTERISK-26689 #close ASTERISK-29929 #close Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb	4 years ago
Luke Escude	5875c7bb6c	res_pjsip_sdp_rtp.c: Support keepalive for video streams. ... ASTERISK-28890 #close Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1	4 years ago
George Joseph	921ab52cf3	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	4 years ago
Florentin Mayer	dd41572f99	res_pjsip_sdp_rtp: Preserve order of RTP codecs ... The ast_rtp_codecs_payloads functions do not preserve the order in which the payloads were specified on an incoming SDP media line. This leads to a problem with the codec negotiation functionality, as the format capabilities of the stream are extracted from the ast_rtp_codecs. This commit moves the ast_rtp_codec to ast_format conversion to the place where the order is still known. ASTERISK-28863 ASTERISK-29320 Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25	4 years ago
Alexander Traud	f6df28ce87	res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites. ... res_sdp_crypto_parse_offer(.) emits many log messages already. ASTERISK-29785 Change-Id: I1a191ebe4fec1102946d4e31887e5197ca02dfe8	4 years ago
Alexander Traud	463f6c83e8	res_pjsip: Fix for Doxygen. ... ASTERISK-29747 Change-Id: Ic7a1e9453f805a6264fe86c96b7d18b87b376084	4 years ago
Bernd Zobl	c30f68a57b	res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress ... With the fix for ASTERISK_28754 channels are no longer put on hold if an outbound INVITE is answered with a "Session Progress" containing "inactive" audio. The previous change moved the evaluation of the media attributes to `negotiate_incoming_sdp_stream()` to have the `remotely_held` status available when building the SDP in `create_outgoing_sdp_stream()`. This however means that an answer to an outbound INVITE, which does not traverse `negotiate_incoming_sdp_stream()`, cannot set the `remotely_held` status anymore. This change moves the check so that both, `negotiate_incoming_sdp_stream()` and `apply_negotiated_sdp_stream()` can do the checks. ASTERISK-29479 Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75	4 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	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
Holger Hans Peter Freyther	9c0ded6e76	res_pjsip_sdp_rtp: Fix accidentally native bridging calls ... Stop advertising RFC2833 support on the rtp_engine when DTMF mode is auto but no tel_event was found inside SDP file. On an incoming call create_rtp will be called and when session->dtmf is set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without looking at the SDP file. Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND but continued to advertise RFC2833 support. This meant the native_rtp bridge would falsely consider the two channels as compatible. In addition to changing the DTMF mode we now set or remove the AST_RTP_PROPERTY_DTMF. The property is checked in ast_rtp_dtmf_compatible and called by native_rtp_bridge_compatible. ASTERISK-29051 #close Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287	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
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
Joshua C. Colp	34750d2068	res_pjsip_sdp_rtp: Only do hold/unhold on default audio stream. ... When examining a stream to determine hold/unhold information we only care about the default audio stream. Other streams aren't used for hold/unhold. ASTERISK-28784 Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4	6 years ago
Michael Neuhauser	5562fb2ea0	chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active ... Do not hang up a PJSIP channel on RTP timeout if that channel is in a direct-media bridge. Also reset the time of the last received RTP packet when direct-media ends (wait full rtp_timeout period before checking first time after audio came back to Asterisk). ASTERISK-28774 Reported-by: Michael Neuhauser Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1	6 years ago
George Joseph	0cde95ec89	Merge "res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated"	6 years ago
Torrey Searle	14ba1806f3	res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated ... If ICE support is enabled but not negotiated, the rtp->ice structure is not being destroyed. This leads to Asterisk waiting for ICE to complete instead of immediately starting the DTLS handshake, resulting in the call leg having no RTP. ASTERISK-28769 #close Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27	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
Torrey Searle	77c9ba8e63	res/res_pjsip_sdp_rtp: Fix MOH transitions ... Update the state of remote_hold immediately on receipt of remote SDP so that the information is available when building the SDP answer ASTERISK-28754 #close Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f	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
Torrey Searle	bf4340f0ec	res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough ... When moh_passthrough is used, asterisk is only generating invites of type sendonly and sendrecv instead of taking fully into account the on hold state of the local and remote parties ASTERISK-28738 #close Change-Id: Iaaad9fbc033cb14803d433b8a4071bc337047761	6 years ago
Kevin Harwell	b805e1237d	srtp: Fix possible race condition, and add NULL checks ... Somehow it's possible for the srtp session object to be NULL even though the Asterisk srtp object itself is valid. When this happened it would cause a crash down in the srtp code when attempting to protect or unprotect data. After looking at the code there is at least one spot that makes this situation possible. If Asterisk fails to unprotect the data, and after several retries it still can't then the srtp->session gets freed, and set to NULL while still leaving the Asterisk srtp object around. However, according to the original issue reporter this does not appear to be their situation since they found no errors logged stating the above happened (which Asterisk does for that situation). An issue was found however, where a possible race condition could occur between the pjsip incoming negotiation, and the receiving of RTP packets. Both places could attempt to create/setup srtp for the same rtp instance at the same time. This potentially could be the cause of the problem as well. Given the above this patch adds locking around srtp setup for a given rtp, or rtcp instance. NULL checks for the session have also been added within the protect and unprotect functions as a precaution. These checks should at least stop Asterisk from crashing if it gets in this situation again. This patch also fixes one other issue noticed during investigation. When doing a replace the old object was freed before creating the replacement. If the new replacement object failed to create then the rtp/rtcp instance would now point to freed srtp data which could potentially cause a crash as well when the next attempt to reference it was made. This is now fixed so the old srtp object is kept upon replacement failure. Lastly, more logging has been added to help diagnose future issues. ASTERISK-28472 Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc	6 years ago
Kevin Harwell	93936e367d	res_pjsip_sdp_rtp: Remove unused variable ... The variable 'endpoint_caps' in function 'set_caps' is not used, so remove. ASTERISK-28458 Change-Id: Ia8766d05a0738aecb29dd018302c2dafca5cab34	6 years ago
Joshua Colp	6bb70c93f1	rtp: Add support for transport-cc in receiver direction. ... The transport-cc draft is a mechanism by which additional information about packet reception can be provided to the sender of packets so they can do sender side bandwidth estimation. This is accomplished by having a transport specific sequence number and an RTCP feedback message. This change implements this in the receiver direction. For each received RTP packet where transport-cc is negotiated we store the time at which the RTP packet was received and its sequence number. At a 1 second interval we go through all packets in that period of time and use the stored time of each in comparison to its preceding packet to calculate its delta. This delta information is placed in the RTCP feedback message, along with indicators for any packets which were not received. The browser then uses this information to better estimate available bandwidth and adjust accordingly. This may result in it lowering the available send bandwidth or adjusting how "bursty" it can be. ASTERISK-28400 Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc	7 years ago
George Joseph	8f9ffe5905	res_pjsip_sdp_rtp: Fix return code from apply_negotiated_sdp_stream ... apply_negotiated_sdp_stream was returning a "1" when no joint capabilities were found on an outgoing call instead of a "-1". This indicated to res_pjsip_session that the handler DID handle the sdp when in fact it didn't. Without the appropriate setup, a subsequent media frame coming in would have an invalid stream_num and cause a seg fault when the stream was attempted to be retrieved. apply_negotiated_sdp_stream now returns the correct "-1" and any media is now discarded before it reaches the core stream processing. ASTERISK-28260 Reported by: Sotiris Ganouris Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f	7 years ago
Joshua Colp	e6b67b2a5d	res_pjsip_sdp_rtp: Allow only single ssrc attribute. ... When processing SSRC attributes we were iterating through all of them, even though we only need to know the remote SSRC once. This was problematic because some browsers group SSRCs together on a stream, and due to our negotiation only end up using the first one. Since we set the second one as the remote SSRC we would drop the received media from them instead of allowing it through. In the future this may be extended to allow SSRC groups and to use information from the attributes. Change-Id: I4dc87087dbe56a83aa65f0f897bbd4ca75ec1270	7 years ago
Xiemin Chen	a526676836	bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix ... To avoid the stream name collide if there're more than one video track in one client. If client has multi video tracks, the name of ast_stream which represents each video track may be the same. Use the MSID:LABEL here because it's identifiable. ASTERISK-28196 #close Reported-by: xiemchen Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b	7 years ago
Joshua Colp	18e206381a	res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled. ... For video streams it was possible for the abs-send-time information to be placed into RTP streams even if not negotiated. Depending on the endpoint in use this could cause video to not flow. We now only enable abs-send-time for negotiation if WebRTC is enabled. ASTERISK-28230 Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c	7 years ago
Joshua Colp	ce9a980be6	pjproject: Upgrade to 2.8. ... This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	7 years ago
Torrey Searle	926d647def	res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered ... If in the initial sdp the caller doesn't include the line a=rtcp-mux Then asterisk shoud not include rtcp-mux in the response regardless of rtcp-mux being enabled on the endpoint ASTERISK-28007 #close Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7	7 years ago
George Joseph	c8e4cd8bce	Merge "res_pjsip_sdp_rtp: include ice in ANSWER only if offered"	7 years ago
George Joseph	34f3fe9552	app_confbridge: Use the SDP 'label' attribute to correlate users ... Previously, the msid "label" attribute was used to correlate participant info but because streams could be reused, the msid wasn't being updated correctly when someone left the bridge and another joined. Now, instead of looking for the msid attribute on a channel's streams, app_confbridge sets an "SDP:LABEL" attribute on the stream which res_pjsip_sdp_rtp looks for. If it finds it, it adds a "label" attribute to the current sdp. Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5	7 years ago
Torrey Searle	1445384699	res_pjsip_sdp_rtp: include ice in ANSWER only if offered ... Keep track if ICE candidates were in the SDP offer & only put them in the corresponding SDP answer if the offer condaind ICE candidates ASTERISK-27957 #close Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92	7 years ago
George Joseph	e7a7506f9c	app_confbridge: Enable sending events to participants ... ConfBridge can now send events to participants via in-dialog MESSAGEs. All current Confbridge events are supported, such as ConfbridgeJoin, ConfbridgeLeave, etc. In addition to those events, a new event ConfbridgeWelcome has been added that will send a list of all current participants to a new participant. For all but the ConfbridgeWelcome event, the JSON message contains information about the bridge, such as its id and name, and information about the channel that triggered the event such as channel name, callerid info, mute status, and the MSID labels for their audio and video tracks. You can use the labels to correlate callerid and mute status to specific video elements in a webrtc client. To control this behavior, the following options have been added to confbridge.conf: bridge_profile/enable_events: This must be enabled on any bridge where events are desired. user_profile/send_events: This must be set for a user profile to send events. Different user profiles connected to the same bridge can have different settings. This allows admins to get events but not normal users for instance. user_profile/echo_events: In some cases, you might not want the user triggering the event to get the event sent back to them. To prevent it, set this to false. A change was also made to res_pjsip_sdp_rtp to save the generated msid to the stream so it can be re-used. This allows participant A's video stream to appear as the same label to all other participants. Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e	7 years ago
Joshua Colp	a507c73a78	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
Ben Ford	f5d5083ea7	res_rtp_asterisk: Add support for receiving and handling NACK requests. ... Adds the ability to receive and handle incoming NACK requests if retransmissions are enabled. If retransmissions are enabled, a data buffer is allocated that stores packets being sent. If a NACK request is received, the packet requested for retransmission is sent if it is still in the buffer. In the same request, if any of the following 16 packets are marked as not received, those will be sent as well if available, as outlined in RFC4585. Also changes RTCP RR and SR to use media source SSRC instead of packet source SSRC when determining which instance to use for RTCP reports. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements ASTERISK-27806 #close Change-Id: I7f7f124af3b9d5d2fd9cffc6ba8cb48a6fff06ec	8 years ago
Joshua Colp	115939caeb	rtp: Add REMB RTP property and set it on PJSIP video RTP. ... This change adds a property to RTP instances to indicate that REMB support is enabled and that sending/receiving should be passed through. This also enables it on video RTP instances in PJSIP if WebRTC support is enabled. Finally the goog-remb extension is added to the SDP using the rtcp-fb attribute to indicate our support for it. Details about REMB can be found on the draft document for it: https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 Change-Id: I1902dda1c0882bd1a0d71b2f120684b44b97e789	8 years ago
Ben Ford	0be1c388e4	Add extended properties to rtp_engine for RTP retransmission support. ... A couple of additional properties are needed in rtp_engine to enable support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically if an endpoint has the webrtc option enabled. While this adds no functionality currently, it will serve as a building block for future changes for RTP retransmission support. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc	8 years ago
Corey Farrell	527cf5a570	Remove redundant module checks and references. ... This removes references that are no longer needed due to automatic references created by module dependencies. In addition this removes most calls to ast_module_check as they were checking modules which are listed as dependencies. Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e	8 years ago
Corey Farrell	9cfdb81e91	loader: Add dependency fields to module structures. ... * Declare 'requires' and 'enhances' text fields on module info structure. * Rename 'nonoptreq' to 'optional_modules'. * Update doxygen comments. Still need to investigate dependencies among modules I cannot compile. Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf	8 years ago