143 Commits (5e6b1476a087407a052f007d326c504cfeefebe7)

Author	SHA1	Message	Date
server-pandora	5e6b1476a0	res_rtp_asterisk.c: Fix DTLS negotiation delays. ... - Trigger pending DTLS packets to send out, once the RTP instance's remote address is set. - Avoids locking the DTLS structure unnecessarily by only doing this if DTLS is passive. - Add DTLS locks around the structurally sensitive calls in the SSL portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock inside of itself, and we're dealing with the SSL BIO in at least two threads. WebRTC channels may receive a DTLS handshake before ast_rtp_remote_address_set is called, which causes there to be a pending response to send out. Previous to 1ad827, this was handled by calling dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP packet could trigger the pending handshake response. Since that was rightfully removed, whenever the DTLS handshake is received before the remote address is set, we would have to wait until another SSL packet arrives. As of Chrome M47's optimizations to their handshake process, WebRTC conversations between Chrome M47+ and Asterisk, where Asterisk is passive, experience a 1 second delay without this patch, because the SSL handshake is received before ICE negotation stores the remote_address, and the next SSL packet isn't received until after a 1 second timeout in Chrome, which causes a new handshake request. ASTERISK-25614 #close Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908	10 years ago
Steve Davies	e74110188d	Further fixes to improper usage of scheduler ... When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b	10 years ago
Matt Jordan	44efdbd2de	res/res_rtp_asterisk: Fix assignment after ao2 decrement ... When we decide we will no longer schedule an RTCP write, we remove the reference to the RTP instance, then assign -1 to the stored scheduler ID in case something else comes along and wants to see if anything is scheduled. That scheduler ID is on the RTP instance. After 60a9172d7e was merged to fix the regression introduced by 3cf0f29310, this improper assignment on a potentially destroyed object started getting tripped on the build agents. Frankly, this should have been crashing a lot more often earlier. I can only assume that the timing was changed just enough by both changes to start actually hitting this problem. As it is, simply moving the assignment prior to the ao2 deference is sufficient to keep the RTP instance from being referenced when it is very, truly, aboslutely dead. (Note that it is still good practice to assign -1 to the scheduler ID when we know we won't be scheduling it again, as the ao2 deref *may* not always destroy the ao2 object.) ASTERISK-25449 Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7	10 years ago
Matt Jordan	60a9172d7e	Fix improper usage of scheduler exposed by 5c713fdf18 ... When 5c713fdf18 was merged, it allowed for scheduled items to have an ID of '0' returned. While this was valid per the documentation for the API, it was apparently never returned previously. As a result, several users of the scheduler API viewed the result as being invalid, causing them to reschedule already scheduled items or otherwise fail in interesting ways. This patch corrects the users such that they view '0' as valid, and a returned ID of -1 as being invalid. Note that the failing HEP RTCP tests now pass with this patch. These tests failed due to a duplicate scheduling of the RTCP transmissions. ASTERISK-25449 #close Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39	10 years ago
Joshua Colp	fa0985851a	res_rtp_asterisk: Move "Set role" warning to be debug. ... In practice the set_role API callback can be invoked even when no ICE is present on an RTP instance. This can occur if ICE has not been enabled on it. ASTERISK-25438 #close Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69	10 years ago
David M. Lee	819760baec	res_rtp_asterisk: Add more ICE debugging ... In working through a recent ICE negotiation bug, I found the debug logging in res_rtp_asterisk to be lacking. This patch adds a number of debug and warning statements that were helpful. Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80	10 years ago
David M. Lee	94b764c8f3	Fix when remote candidates exceed PJ_ICE_MAX_CAND ... We were passing the wrong count into pj_ice_sess_create_check_list(), causing the create to fail if we ever received more than PJ_ICE_MAX_CAND candidates. Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378	10 years ago
Joshua Colp	c7a1dca4ba	res_rtp_asterisk: Don't leak temporary key when enabling PFS. ... A change recently went in which enabled perfect forward secrecy for DTLS in res_rtp_asterisk. This was accomplished two different ways depending on the availability of a feature in OpenSSL. The fallback method created a temporary instance of a key but did not free it. This change fixes that. ASTERISK-25265 Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396	10 years ago
Mark Duncan	2d2e741905	res/res_rtp_asterisk: Add ECDH support ... This will add ECDH support to Asterisk. It will detect auto ECDH support in OpenSSL (1.0.2b and above) during ./configure. If this is available, it will use it, otherwise it will fall back to prime256v1 (this behavior is consistent with other projects such as Apache and nginx). This fixes WebRTC being broken in Firefox 38+ due to Firefox now only supporting ciphers with perfect forward secrecy. ASTERISK-25265 #close Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b	10 years ago
Joshua Colp	47ebab959e	res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used. ... This change fixes a bug where the DTLS timeout timer would be initialized to 0 if DTLS was not used for an RTP session. ASTERISK-25103 Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac	10 years ago
Joshua Colp	1ad827327a	res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context. ... This change moves logic for setting up the DTLS SSL contexts to when the SDP is done being processed instead of when ICE negotiation completes. It also stops handshakes from being initiated when we are acting as a server. Manipulating the SSL context when ICE negotiation has completed is problematic as the SSL context is not protected and if acting as a client the remote side may have started DTLS negotiation already. The retransmission timeout timer code has also been split up and simplified some. Both RTP and RTCP now have their own timers and the points at which the timer is stopped and started is now more specific. When a packet is sent the timer is started. When a response is received but before it is processed the timer is stopped. This provides a guarantee that the timeout is not occurring while the response is processed. ASTERISK-22805 #close ASTERISK-24550 #close ASTERISK-24651 #close ASTERISK-24832 #close ASTERISK-25103 #close ASTERISK-25127 #close Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91	10 years ago
Yousf Ateya	e3129b84b1	res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS. ... First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values. Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31	10 years ago
Sean Bright	151c3316d5	res_rtp_asterisk: Issue ERROR if res_srtp is not found. ... While trying to get WebRTC working with chan_pjsip, I was running into the following error: Attempted to set an invalid DTLS-SRTP configuration on RTP instance... Josh helpfully pointed out that res_srtp.so might not be loaded, and sure enough, it wasn't. This patch adds a ERROR indiciating as much to hopefully help others having a similar problem. Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f	10 years ago
Steve Davies	a9b25f035e	res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS ... ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created. The resources are linked into a table, but the original alloc refs are never released. ast_strdup leak in rtp_engine.c. If ast_rtp_dtls_cfg_copy() is called twice on the same destination struct, a pointer to an alloc'd string is overwritten before the string is free'd. ASTERISK-25022 Reported by: one47 Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b	10 years ago
Matthew Jordan	297b8df31b	clang compiler warnings: Fix -Wabsolute-value warnings ... This patch fixes several warnings caught by clang - in this case, usage of the abs function on non-integer values. This patch uses labs and fabs, as appropriate, in the various affected files. Review: https://reviewboard.asterisk.org/r/4525 ASTERISK-24917 Reported by: dkdegroot patches: rb4525.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	9fe6588349	res_rtp_asterisk: Fix DTLS when used with OpenSSL 1.0.1k ... A recent security fix for OpenSSL broke DTLS negotiation for many applications. This was caused by read ahead not being enabled when it should be. While a commit has gone into OpenSSL to force read ahead on for DTLS it may take some time for a release to be made and the change to be present in distributions (if at all). As enabling read ahead is a simple one line change this commit does that and fixes the issue. ASTERISK-24711 #close Reported by: Jared Biel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c0c395514c	chan_dahdi.c, res_rtp_asterisk.c: Change some spammy debug messages to level 5. ... ASTERISK-24337 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	822abf9e9b	Fix printf problems with high ascii characters after r413586 (1.8). ... In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) *ptr); +out += sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	6c969a115c	res_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't. ... In the case where the ICE negotiation had not yet started current state would get wiped when it shouldn't. This also removes channel binding as in practice this does not work well with other implementations. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b69057cd76	res_rtp_asterisk: Make the ICE transport check case insensitive as some implementations use 'udp'. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	6b2ee3002f	res_rtp_asterisk: Crash if no candidates received for component ... When starting ice if there is not at least one remote ice candidate with an RTP component asterisk will crash. This is due to an assertion in pjnath as it expects at least one candidate with an RTP component. Added a check to make sure at least one candidate contains an RTP component and at least one candidate has an RTCP component. ASTERISK-24383 #close Review: https://reviewboard.asterisk.org/r/4039/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	89247cec8e	res_rtp_asterisk: Allow only UDP ICE candidates. ... The underlying library, pjnath, that res_rtp_asterisk uses for ICE support does not have support for ICE-TCP. As candidates are passed through directly to it this can cause error messages to occur when it receives something unexpected (such as a TCP candidate). This change merely ignores all non-UDP candidates so they never reach pjnath. ASTERISK-24326 #close Reported by: Joshua Colp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	fd4191e016	res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP. ... This change fixes an issue where ICE candidates put into the SDP did not contain the 'raddr' and 'rport' information for server reflexive and relay candidates. #SIPit31 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0ea0c38600	res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	38885ac8bd	res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning. ... Side note: I need a vacation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	222d6ac4ab	res_rtp_asterisk: Fix building when pjproject is not used. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ff4371e45d	res_rtp_asterisk: Fix a myriad of TURN client issues. ... 1. The number of file descriptors an ioqueue instance can handle is fixed, so we now spawn the required number to handle the load. 2. Our transport identifiers were exceeding the range supported by pjnath. 3. The TURN client did not set up client binding causing needless bandwidth usage. 4. The code no longer updates address information on each packet. 5. STUN traffic was getting looped back to Asterisk instead of going through the TURN server. 6. Synchronization now ensures things are completely setup or destroyed. 7. Logging now reflects the target the TURN server is sending to/receiving from on our behalf. ASTERISK-23577 #close Reported by: Jay Jideliov ASTERISK-23634 #close Reported by: Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	915de454f8	res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. ... This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3e642e66d1	res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0. ... ASTERISK-23834 #close Reported by: Richard Kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abac3330cf	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	374f32fcc9	res_rtp_asterisk: Add support for DTLS handshake retransmissions ... On congested networks, it is possible for the DTLS handshake messages to get lost. This patch adds a timer to res_rtp_asterisk that will periodically check to see if the handshake has succeeded. If not, it will retransmit the DTLS handshake. Review: https://reviewboard.asterisk.org/r/3337 ASTERISK-23649 #close Reported by: Nitesh Bansal patches: dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	7e083b9781	res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE ... ICE sessions will now be restarted if sessions are changed to use new sets of remote candidates. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c382714769	res_rtp_asterisk: correct build error from r409129 ... Accidentally placed a declaration below functional code (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	8a7ef07f94	res_rtp_asterisk: Fix checklist creating problems in ICE sessions ... Prior to this patch, local candidate lists including SRFLX would fail to start properly when building ICE candidate check lists. This patch fixes that problem by making sure that each SRFLX candidate is associated with the proper base address so that the check list can create matches properly. This patch was written by jcolp. The issue will be left open to await testing by the issue participants. (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	1968d29447	res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions ... ast_bind to a port reserved for another program by SELinux causes errno == EACCES. This caused random failures when binding rtp or udptl sockets. Treat EACCES as a non-fatal error, try next port. (closes issue ASTERISK-23134) Reported by: Corey Farrell ........ Merged revisions 406933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	dee9ce4a0b	res_rtp_asterisk: Fails to resume WebRTC call from hold ... In ast_rtp_ice_start if the ice session create check list failed, start check was never initiated and ice_started was never set to true. Upon re-entering the function (for instance, [un]hold) it would try to create the check list again with duplicate remote candidates. Fixed so that if the create check list fails the necessary data structures are properly re-initialized for any subsequent retries. Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a check list failure because it possible things might still work. However, a debug message was added to help with any future troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Patches: works_on_my_machine.patch uploaded by xytis (license 6558) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2d5debf45c	chan_sip: Fix RTCP port for SRFLX ICE candidates ... This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4241f7b6ea	res_rtp_asterisk: Address jittery DTMF events in RTP streams ... (closes issue ASTERISK-21170) Reported by: NITESH BANSAL Patches: dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418) Review: https://reviewboard.asterisk.org/r/2938/ ........ Merged revisions 401619 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	0336f4b606	res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change ... In r400089, a patch was put in to correct erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can be called on an RTP instance that does not have RTCP information. This patch prevents that crash by only resetting the statistics if we do actually have an RTCP instance. (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John Bigelow ........ Merged revisions 401445 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	69ab5a7cd1	Fix STUN crash when using IPv6 any address ... Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4 candidates are also added. (closes issue ASTERISK-21917) Reported by: Torrey Searle Patches: 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ecdd1e76eb	res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports ... RTCP's calculation of the number of lost packets in an RTP stream is based on that stream's sequence number count, the number of received packets, and how many packets we expect to receive. When the SSRC for an RTP stream changes, there can - and almost always will be - a large jump in the next packet's timestamp and sequence number. If we don't reset the number of received packets, sequence number count, and other metrics used by RTCP, the next RR/SR report will use the previous SSRC's values to calculate the lost packet count for the new SSRC - resulting in a very large number of lost packets. This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it will reset the various values used by the RTCP calculations. From the perspective of RTCP, this appears as a new media stream - which is what it is. Review: https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174) Reported by: Thomas Arimont ........ Merged revisions 400089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	71ce810908	Make libuuid an optional dependency for res_rtp_asterisk instead of a requirement. ... Review: https://reviewboard.asterisk.org/r/2777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	08c2a533f2	Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit ... When we send out a CN packet (for instance, in the case of using rtpkeepalives), we are not setting the payload code properly. Also, we are setting the marker bit when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup(). 11 and trunk already use the appropriate function. * In 1.8, use ast_rtp_codecs_payload_code() * Remove the setting of the marker bit * Fix the debug message by incrementing the seqno after the debug message is set in order to display the correct seqno that was sent out (closes issue ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter Katzmann, Michael L. Young Patches: asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2500/ ........ Merged revisions 388111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a1d8e4fbd6	Clear the DTMF sending digit tracking on off nominal paths ... In certain situations, when the RTP engine goes to send a DTMF end digit it may be in a situation where the remote address is no longer available, or the digit that was supposed to be sent is invalid. In such cases, we need to clear the RTP counters appropriately. Otherwise, when the RTP source is set again, we'll continue to think that we're in the middle of sending a DTMF digit, which can confuse the remote party (signficantly). (closes issue ASTERISK-21522) Reported by: Corey Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909) ........ Merged revisions 387213 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	d8216bd9ee	Add dependency on libuuid, for res_rtp_asterisk ... pjproject is what actually requires libuuid. (closes issue ASTERISK-21125) reported by Private Name (Ed. note: Really? Private Name? I am rolling my eyes so hard right now.) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d014e51a18	Fix white noise on SRTP decryption ... When res_rtp_asterisk.c was altered to avoid attempting to apply unprotect algorithms to non-audio RTP packets, the test used was incorrect. This caused the audio packets to not be decrypted and resulted in loud white noise on the other endpoint (or both endpoints depending on the call legs involved). The test now properly checks the version field in the RTP header to ensure that RTP and RTCP are decrypted while other types of packets are not. (closes issue ASTERISK-21323) Reported by: andrea Tested by: Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff uploaded by Kinsey Moore ........ Merged revisions 384048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5b1533ca7b	Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source ... Often, Asterisk may realize that a change in the source of an RTP stream is about to occur and ask that the RTP engine reset it's lock on the current RTP source. In certain scenarios, it may take awhile for the new remote system to send RTP packets, while the old remote system may continue providing RTP during that time period. This causes Asterisk to re-lock onto the old source, thereby rejecting the new source when the old source stops sending RTP and the new source begins. This patch prevents that by having a constant secondary, 'secret' probation mode enabled when an RTP source has been chosen. RTP packets from other sources are always considered, but never chosen unless the current RTP source stops sending RTP. Review: https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124) Reported by: John Bigelow Tested by: John Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	a78bb96d94	While the ICE negotiation is occurring leave strictrtp in an open state, media can and will come from different places. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	f6b368216a	Fix a bug with ICE and strictrtp where media could get dropped. ... If the end result of the ICE negotiation resulted in the path for media changing it was possible for the strictrtp code to discard the RTP packets. This change causes strictrtp to enter learning mode once again when the ICE negotiation has completed successfully. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	af1666f1cf	Reset RTP timestamp; sequence number on SSRC change ... In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to better account for out of order RTP packets. This was accomplished by using the RTP timestamp and sequence number to check for out of order packets. However, when a SSRC change occurs, the timestamp and sequence number will no longer have any relation to the previously received packets. The variables tracking the timestamp and sequence number therefore have to be reset. (closes issue ASTERISK-20906) Reported by: Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442) ........ Merged revisions 378967 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago