3205546bcb
CID:1210582: remove logically dead code
2015-05-28 12:47:24 -05:00
1550d548db
CID:1024555: remove logically dead code
2015-05-28 12:47:24 -05:00
c6bd6aea4e
FS-7499: juggle log lines
2015-05-28 12:47:20 -05:00
3e24ac5e6b
FS-7501: add auto sync of jb and fps detection
2015-05-28 12:47:18 -05:00
c8a189a433
FS-7499: demote log line
2015-05-28 12:47:18 -05:00
42e7b81b1e
FS-7500 FS-7508: move debug logging to DEBUG1
2015-05-28 12:47:17 -05:00
d293e9bd1b
FS-7500: check for uninit srtp
2015-05-28 12:47:17 -05:00
d5e48302e6
FS-7501: improve linked list algorithm in a few places to help performance
2015-05-28 12:47:17 -05:00
fa7695847a
FS-7499: improve generic nack and vpx framing
2015-05-28 12:47:14 -05:00
7c294f242f
FS-7504: allow <modname>.<codecname> support so multiple modules can exist for the same codec
2015-05-28 12:47:13 -05:00
d418fb37ed
FS-7500: init dtmf to 0
2015-05-28 12:47:11 -05:00
24254bb1fd
FS-7500: revert
2015-05-28 12:47:11 -05:00
d3359ff9f0
FS-7500: don't wait for video ready from inside video thread that sets that flag
2015-05-28 12:47:11 -05:00
1cd9e52b9e
FS-7499: add enum for various rtcp related types
2015-05-28 12:47:10 -05:00
3e7c0f6558
FS-7499: fix seg
2015-05-28 12:47:09 -05:00
6e05e09e9a
FS-7500: missing newline
2015-05-28 12:47:09 -05:00
0d34e8ac77
FS-7500: add a framebuffer to reuse memory and use it to offload frame writing from video muxing thread to a dedicated write thread
2015-05-28 12:47:08 -05:00
eb78d2ae7b
FS-7499: ignore replay errs when nack is enabled
2015-05-28 12:47:07 -05:00
59da14542f
FS-7505 FS-7514: working towards vid rec
2015-05-28 12:47:06 -05:00
2c1ab14074
FS-7513: add configurable FPS for conf and default to 15
2015-05-28 12:47:02 -05:00
8a1cb14015
FS-7499: trying to fix rtp data len when rtp extension is used
...
duplicated some code from 4943~4953, but that code has it's own problem, it forget to reset *bytes results to
larger frame->datalen could read beyond the buffer, and it also makes stats not accurate. But if we reset *bytes
at that place, then later the switch_vb_put_packet has problem because it depends that *bytes. this patch should
fix the datalen at least buf still leaves duplicated code and inaccurate stats.
2015-05-28 12:46:59 -05:00
fa5d6af2cd
FS-7513: refactor conference video muxing to create one distinct encoder per codec used and only create one encoded frame per distinct codec, store current image used by layer on the layer so it is not destroyed before the canvas is written, refactor and rearrange some functions
2015-05-28 12:46:57 -05:00
d6ef34a725
FS-7508: trying to mitigate chrome going crazy on reload
2015-05-28 12:46:55 -05:00
4d100bc2e8
FS-7509: stop media on verto detach
2015-05-28 12:46:55 -05:00
59fa1b9ac7
FS-7499: mod vid refresh stuff
2015-05-28 12:46:55 -05:00
f110ce40e2
FS-7501: mod of video i/o for jb
2015-05-28 12:46:54 -05:00
2983c7e6df
FS-7499: keep track of from addr from rtp separate since on ice you get stun packets etc from other ip. This helps auto adjust work properly
2015-05-28 12:46:54 -05:00
a006d53a3d
FS-7499: tweak nack and fir handling
2015-05-28 12:46:54 -05:00
81887e9bfc
FS-7501: add video jitterbuffer debug controls
2015-05-28 12:46:53 -05:00
ac2e1b692e
FS-7501: tweak some settings on jb
2015-05-28 12:46:53 -05:00
17aa836403
FS-7499: add generic nack support to rtp stack
2015-05-28 12:46:53 -05:00
b63683ade0
FS-7501: more code
2015-05-28 12:46:52 -05:00
0d626bc715
FS-7501: more factoring on vid buffer
2015-05-28 12:46:52 -05:00
0e991e7d0f
FS-7501: connect video buffer for testing, still needs a lot of work
2015-05-28 12:46:52 -05:00
2a50c6d55c
FS-7501: use vidderbuffer in rtp
2015-05-28 12:46:52 -05:00
faa99a7a47
FS-7499: don't send fir or pli till stun is established
2015-05-28 12:46:51 -05:00
ae44bd27e2
FS-7499: tweaks to rate of fir/pli
2015-05-28 12:46:51 -05:00
76ec99ed97
FS-7500: poll rtp on answer until dtls is negotiated
2015-05-28 12:46:51 -05:00
303a4ecf99
FS-7499: move fir and pli into the normal rtcp code so it can be bundled with a report block per the rfc
2015-05-28 12:46:51 -05:00
de4a0e7a3c
FS-7500: nevermind
2015-05-28 12:46:50 -05:00
970064294c
FS-7500: refactoring
2015-05-28 12:46:50 -05:00
b0fd27bb8f
FS-7500: comment debug
2015-05-28 12:46:50 -05:00
91602e9cfa
FS-7499: properly decode rtcp
2015-05-28 12:46:49 -05:00
45898cfad7
FS-7500: better version of last commit
2015-05-28 12:46:49 -05:00
b747687bb2
FS-7500: set ssrc from frame not rtp session so the ssrc changing coded can detect a shift
2015-05-28 12:46:49 -05:00
01fda5748c
FS-7500: another round of trying to make things work
2015-05-28 12:46:48 -05:00
b8ba1a1469
FS-7500: reduce CNG frames on video and move debug from mod_fsv to the core with a flag to enable it since the raw packet is not available anymore when you set DECODED READ flag
2015-05-28 12:46:47 -05:00
140a1c9661
FS-7500 FS-7508: shift some hacks around
2015-05-28 12:46:47 -05:00
73b2a5ea87
FS-7500: tmp comment
2015-05-28 12:46:45 -05:00
0cd5658caa
FS-7500: another refactoring pass, temp code still in place, WORK IN PROGRESS
2015-05-28 12:46:44 -05:00
3c29d4e8a7
FS-7500: mark places to fix later
2015-05-28 12:46:44 -05:00
659c1e474e
FS-7500: Work in progress. Added codec config params that can be set from session and made vpx codec re-init on size change. Also add periodic key frame timer
2015-05-28 12:46:44 -05:00
365a5dd820
FS-7500: major refactoring pass. Push concepts from mod_vlc as deep as possible and flesh out api to use everywhere else. Round 2 will be to convert the bridge and other places using the same code
2015-05-28 12:46:44 -05:00
765fff3d75
FS-7500: add support for codec control and use it to pass messages down to the codec and use it to implement keyframe reset for fir, pli and nack. Later we will expand to handle nack correctly.
2015-05-28 12:46:44 -05:00
1b322bd952
FS-7425: #resolve dhparams might not be present, causing a seg. Make sure they are there before we apply them
2015-05-04 11:23:33 -04:00
c143ef1b3d
FS-7466
2015-04-29 19:18:59 -05:00
b9b1b61d20
FS-7425: set dh params and call set_tmp_dh to enable PFS for DTLS-SRTP
2015-04-24 10:31:17 -05:00
921f1a2bd2
FS-7458
2015-04-21 12:06:42 -05:00
638e932422
FS-7434 reset jitter buffer when SSRC changes
2015-04-16 16:02:16 -04:00
183570bd94
FS-7396: #resolve update dtls socket when socket changes on auto-adjust that changes address families and also include link local v6 addresses in approrpiate auto acls
2015-03-26 17:38:12 -05:00
5e43c6dd25
Merge pull request #170 in FS/freeswitch from ~NIMAST/freeswitch-fs-7203:rtcp-source-fraction-fix to master
...
* commit '5f7e111f79dd1a965aa956da7495485f52b0a1cc':
Fix source fraction always 0 in RTCP events
2015-03-06 13:36:37 -06:00
302a339fdf
FS-7294: Enable -Werror when building with clang compiler #resolve
2015-02-17 12:20:33 -05:00
5f7e111f79
Fix source fraction always 0 in RTCP events
...
Without the value for source fraction, applications relying on RTCP events for making changes to FS behaviour or even for logging get false information.
With this change the value for source fraction is passed along in RTCP events correctly.
To my current understanding, as the value for fraction in the RTCP packet is represented by 8 bits according to the spec, calling `ntohl` on it will always zero it out. Fixed by removing the call.
FS-7203 #resolve
2015-01-27 18:13:04 +02:00
8d599a82bc
one more tweak to not jump back and forth on ice when you have 2 reachable
2015-01-26 15:33:33 -06:00
90d3cb633c
fix media reload on verto and sip re-invites
2015-01-22 03:07:50 -06:00
95a8efb174
up the ice failover val to 3 sec
2015-01-21 01:21:31 -06:00
46cf8a4dce
fix seg in ice rtp code
2015-01-17 00:22:11 -06:00
3e6ffbcf06
FS-7144 #resolve
2015-01-12 18:55:32 -06:00
a2b5356dae
FS-7131 #comment please test
2015-01-09 21:47:28 -06:00
ba016c2850
FS-7095 #comment please test
2014-12-18 13:08:11 -06:00
cee8b30c45
set rtp_has_crypto for dtls calls
2014-12-16 10:19:43 -06:00
e783999b51
some changes to webrtc to make it work with iDoubs in rtcweb profile mode
2014-12-12 20:55:40 -06:00
d1e529aefd
Add new hard_mute control to allow apps to request low level mute e.g. from the rtp stack level. Its used in mod_conference to avoid reading audio while muted and possibly reduce some transcoding load
2014-10-27 15:13:42 -04:00
1f9025d446
FS-6926 #resolve #comment please test and reopen if necessary
2014-10-16 17:57:46 -05:00
6bfc05b81e
FS-6887 #resolve #comment new bug flag always_auto_adjust (also implicitly sets accept_any_packets)
2014-10-02 11:55:53 -05:00
8f85b5204c
vs2010 trival compiler warnings
2014-09-17 18:11:20 -05:00
f924684eff
FS-6623 #resolve fix init and logging for rtcp
2014-09-15 20:08:09 +05:00
b738775876
[FS-6623] implement RTCP report generation
2014-09-15 20:08:09 +05:00
3e8e2ce151
Revert commits pushed too early
...
Revert "depend on fs before install"
This reverts commit 6c522179206df5288f5a0e89bbd033764faad67150c62cdfd7ea973b0b4c0b7863a9b7
2014-09-12 17:07:50 +00:00
6df5288f5a
removing commented work in progress on SDES and logging tunning on
...
rtcp_init
2014-09-12 11:58:54 -05:00
0e89bbd033
more formatting and logging tuning
2014-09-12 11:58:53 -05:00
764faad671
logging adjustment
2014-09-12 11:58:53 -05:00
50c62cdfd7
missing host to network conversion highest_sequence_number_received
2014-09-12 11:58:53 -05:00
ea973b0b4c
logging correction
2014-09-12 11:58:53 -05:00
0b7863a9b7
[FS-6623] implement RTCP report generation
2014-09-12 11:58:53 -05:00
37d7fb7888
calculate jitter percentage in jitterbuffer to factor into conditions for reducing the size when in adaptave mode
2014-09-10 04:17:01 +05:00
151440b7e1
fix race caused by consecutive stun packets
2014-09-09 21:35:51 +05:00
aa1a05d0aa
Help the static analyzer in `handle_ice`
...
Clang's static analyzer thinks we could be using `hosts` here when it
is NULL. We probably weren't, but it's easy to see how it could think
so. We were checking whether `from_addr` matched `ice->addr` three
times, and between the second on third time we might have modified the
`ice->addr`; however we only get there if it matched the second time,
so we could only make it not match at that point and avoid the third
branch. We can't make it match where it did not before.
We'll simplify the logic a bit here so static analyzers (and humans)
can hopefully see this more readily.
2014-08-22 03:37:42 +00:00
3526ca5cb5
Allow setting threshold for RTP auto adjust
...
If we see a certain number of RTP packets from a host and port other
than was negotiated, we adjust to send our RTP to that host and port.
Traditionally we've waited for 10 packets. This commit makes the
threshold adjustable by setting the channel variable
`rtp_auto_adjust_threshold` to any positive value less than 2^16.
2014-07-16 01:32:18 +00:00
aef569172b
Removed a useless called to abs.
...
Clang 3.5 reported the following error: error: taking the absolute
value of unsigned type 'unsigned int' has no effect
[-Werror,-Wabsolute-value]
Subtracting unsigned variables will never be negative and will either
be the small expected value or will wrap to a very big value. This
code is trying to determine if the difference between these timestamps
is greater than 16000.
The variables last_write_ts and this_ts deal with timestamps. In the
normal case this_ts will be a larger timestamp than
last_write_ts. This change will maintain the intended behavior of
reseting the video if the difference is larger than
16000 and in the abnormal case this value would wrap and still exceed
the 16000.
2014-07-03 13:17:12 -07:00
c1f1f8b98e
Check for too many SRTP errors before warning
...
We're checking whether we've hit the warning threshold before checking
whether we should just end the call. This causes an off-by-one error
where we take one SRTP error more than intended.
This commit reverses the order of the tests.
2014-06-29 20:49:46 +00:00
f31641f4bf
Allow more SRTP errors before killing call
...
In a carrier interop we saw the call get killed for SRTP failures
during a reinvite. We're wondering if the SRTP errors may have been
transitory and if it may have recovered after a few more packets.
It's debatable whether we should kill calls at all for SRTP auth
failures; semantically the right thing to do when a MAC fails is to
ignore the packet completely. So raising this limit to 100 packets
shouldn't do any harm. With this change we still warn at 10 errors
and every 10 errors thereafter.
2014-06-28 03:57:20 +00:00
7406be6927
Relay cause of hangup on SRTP failure
...
We hangup the channel after receiving 10 SRTP packets in a row with a
bad auth tag or that are replayed. Prior to this commit we were
indicating a normal clearing. When doing interop and looking first at
packet traces, this made freeswitch's behavior look surprising. With
this commit we'll indicate more loudly what's happening.
2014-06-28 01:18:50 +00:00
52892b312a
Fix misspelled function
...
switch_rtp_set_invalid_handler has been misspelled as
switch_rtp_set_invald_handler going all the way back to the
beginning. So while it's possible that someone somewhere could be
relying on this misspelling, I think it's more likely that no one has
used it much and that's why it wasn't spotted. We don't even use it
ourselves anywhere anymore.
Introduced in commit: 828e03715f
2014-06-28 00:32:41 +00:00
3c08104874
remove unused code
2014-06-18 01:17:35 +05:00
c0e7e7b88c
add reset function to clear some state data in the rtp session
2014-06-14 07:05:00 +05:00
c375e336bc
add debugging
2014-06-13 06:06:14 -04:00
0eda5cb80f
suppress audio flaw tally when coming off hold
2014-06-02 19:09:10 -05:00
b58bbd18b0
CID:1214233 Pointer to local outside scope
2014-05-16 21:08:53 +00:00
be56bbb7ae
let relay work if its the only option
2014-05-09 01:14:52 +05:00
59734d8e15
add bounds check to keep rtcp packets with > 5 report blocks from creating a buffer overrun
2014-04-28 13:32:01 -04:00
59fd9b90d0
Correct display of last write timestamp
...
On start DTMF packets we were showing the last write timestamp as a
signed value when it's an unsigned value, which could result in it
appearing incongruous with later packets where the value was displayed
correctly.
2014-04-19 01:48:49 +00:00
7151d6acea
FS-6402 part 2
2014-04-02 03:21:37 +05:00
5c0cff70b3
FS-6402 --resolve
2014-04-02 01:20:19 +05:00
aa147fa5fd
FS-6412 --resolve
2014-03-31 16:22:33 -05:00
087b2e4f30
revert part of 390e6713cc
2014-03-10 14:42:52 -05:00
804ef7709d
change from sqlite hash to newly added one
2014-03-09 00:37:17 +05:00
a491df05f1
declinatio mortuus obfirmo!
2014-03-07 03:35:36 +05:00
390e6713cc
part of last patch
2014-03-07 02:59:09 +05:00
e9847afe22
feed all packets to jitterbuffer when enabled to absorb bursts and improve smoothing and delay protection
2014-03-07 02:48:56 +05:00
e5b291514c
FS-5755
...
rtp_secure_media=mandatory
rtp_secure_media=optional
rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
rtp_secure_media=forbidden
true implies mandatory
false implies forbidden
not set implies optional
rtp_secure_media_inbound or rtp_secure_media_outbound take precedence and are treated the same way based on leg direction
2014-03-06 07:34:47 +05:00
411a76020a
Improve channel variable name to srtp_allow_idle_gaps
...
This was momentarily called force_send_silence_when_idle, but that was
non-obvious as you had to set that value to true to be able to not
send silence when idle. This name describes the purpose much better.
2014-03-04 01:51:04 +00:00
5a7ea956b9
Add force_send_silence_when_idle channel variable
...
If set to true, this prevents us from overriding the value of
send_silence_when_idle. When that is unset or set to zero and SRTP is
engaged, we typically override the value because many devices can't
handle gaps in the SRTP stream.
This variable is mostly for testing whether particular devices can
handle this behavior. Use at your own risk.
2014-03-04 00:09:02 +00:00
20da552564
Preserve value of send_silence_when_idle if possible
...
In commit 55d01d3def
2014-03-03 23:43:29 +00:00
719850e508
FS-5895 --resolve
2014-03-01 04:55:04 +05:00
55d01d3def
Send silent packets when idle with SRTP
...
Originally we did the same thing with SRTP that we do without SRTP,
which is to simply not send packets when e.g. sleep is called.
At commits d63323977f5259814aee
2014-02-28 23:13:37 +00:00
8cee05987e
check the jitter stats after the jitter buffer when its enabled
2014-03-01 02:50:17 +05:00
7b5d17802f
FS-6268 usinga macro to find the rtp_session_name its better on the eyes
2014-02-27 10:42:43 -06:00
378caebc9a
fix --disable-srtp
2014-02-26 08:05:22 -06:00
5646957c5b
FS-5937
2014-02-26 04:06:59 +05:00
463f32c4e3
FS-5937: i need to build a test rig for this, go go gadget iphone commit
2014-02-24 23:44:44 -06:00
7aff64b2d2
fix compiler warning vs2010
2014-02-24 23:29:15 -06:00
780890b5de
FS-6240 --resolve
2014-02-24 17:06:01 -05:00
a900eadf5b
FS-5937 --resolve
2014-02-24 14:56:49 -06:00
97e234048c
FS-6204 refactor some dtls code to generate a 1024 bit cert suitable for dtls in all version of chrome
2014-02-19 07:05:37 +05:00
6e7d5d0897
update copyright header for 2014
2014-02-12 12:08:56 -06:00
03c723cfd1
FS-6195 --resolve
2014-02-06 22:24:05 +05:00
5228e02b19
FS-5544 after going over this with a fine tooth comb, all I could find was some issues behind asymetrical nat. If there is a real problem here, post the flipping logs I asked for days ago.......
2014-02-04 11:56:30 +05:00
59d10edd79
Fix build on x86/i386 systems
2014-01-31 17:49:44 +00:00
7b44e5b27c
compiler warnings
2014-01-29 20:46:38 -06:00
4591088971
remove extra debug
2014-01-30 05:37:05 +05:00
32b2a45efc
add more rtp statistics with several new vars for cdr processing in_jitter_min_variance in_jitter_max_variance in_jitter_loss_rate in_jitter_burst_rate in_mean_interval in_flaw_total in_quality_percentage in_mos
2014-01-30 05:08:34 +05:00
0e68325319
only send cng when its setup
2014-01-25 03:51:27 +05:00
358b5db0e4
block all possible routes to write data during ice and dtls negotiation
2014-01-25 03:48:32 +05:00
6fef880f12
FS-5675 --resolve
2014-01-24 00:23:58 +05:00
084e245085
Lots of trivial tweaks to minimize compiler warnings, especially on 64-bit platform: Added missing type casts, changed types, added missing NULL checks.
2014-01-22 22:30:23 +01:00
8dfbd91a3d
fold version macros into runtime functions to avoid cascade rebuilds when modifying the core
2014-01-17 01:54:47 +05:00
f4a4ba1716
FS-5544 --resolve
2014-01-15 21:20:06 +05:00
1b9c21ee88
some webrtc tweaks
2014-01-15 07:04:24 +05:00
e71dabae50
FS-6007
2013-12-31 00:21:51 +05:00
910c661508
fix trivial compiler warning for last commit
2013-12-27 14:52:52 -06:00
de65c19c3c
FS-6007 FS-5083 FS-4514 --resolve
2013-12-28 01:39:41 +05:00
6c5a17894c
accept any payload when no payload map is present
2013-12-05 16:35:49 -06:00
2747660269
FS-5959 FS-5958
2013-11-25 12:08:17 -06:00
b8dc07132a
FS-5959 i hate this ticket
2013-11-22 08:55:45 +05:00
c07a6b6b9e
add polycom support to multicast paging, broadcast all formats at once to support most every popular phone at once
2013-11-22 04:52:11 +05:00
ccaa3ae732
FS-5959
2013-11-21 01:38:21 +05:00
a4dd6ccb42
fix some issues in passthrough of fec packets
2013-11-09 23:02:11 +05:00
bb9adff511
rework media code
2013-11-08 03:48:00 +05:00
2589bf7750
minor webrtc tweaks
2013-11-02 01:21:35 +05:00
b0d95efd09
dont byteswap the extended bit headers in the actual packet so it can properly echo back without modification
2013-10-31 17:22:09 -05:00
fd97365d4a
detect ca-bundle.crt for chain cert to use in dtls connections
2013-11-01 02:22:11 +05:00
88a13f1fc2
update err msg
2013-10-29 13:40:27 -05:00
562ef81dd9
fix webrtc rtcp issue on reinvite
2013-10-29 03:44:07 +05:00
ce83817318
webrtc tweaks
2013-10-28 23:39:33 +05:00
6a7793f72a
change dtmf vol to 13
2013-10-17 12:33:55 -04:00
af4c91764d
only test for reneg after 20 sec with no activity
2013-10-17 03:58:09 +05:00
4a172402d2
some refactoring and plumbing for 1.4
2013-10-16 02:24:32 +05:00
850c154bd4
fix vid switching in webrtc
2013-10-15 04:57:57 +05:00
c0dc8ed6ff
deal with rtp keep alives
2013-10-07 11:38:08 -05:00
d9216daedb
Fix regression from ignoring breaks in video rtp sessions and add a flag to force a break when needed
2013-09-12 21:47:54 +05:00
8def066077
FS-5774 --resolve
2013-09-11 03:01:45 +05:00
5b4b1dce71
FS-5754
2013-09-04 14:15:44 -05:00
952c37d940
FS-5296 --resolve
2013-08-22 02:30:59 +05:00
ef62351c47
FS-5693 --resolve
2013-08-14 02:00:36 +05:00
f7e6c6d6c0
FS-5670 --resolve allow debug rtp if rtp is not attached to a session
2013-08-08 08:22:00 +08:00
39ad799683
FS-4932 FS-5090 --resolve
2013-07-29 10:18:05 -05:00
8253bab19d
send 2 firs to make sure it works
2013-07-17 12:46:10 -05:00
eebac2ea95
webrtc video tweaks
2013-07-16 19:16:01 -05:00
4bf4bf83a8
tweak
2013-07-12 16:59:35 -05:00
92452ad309
fix windows compiler error
2013-07-12 14:27:45 -05:00
5dbf2b3cf7
refactor some video code
2013-07-11 17:38:24 -05:00
8de63a750e
FS-5571 --resolve
2013-07-08 10:02:38 -05:00
e1ed386bb9
FS-5565 --resolve
2013-07-01 12:16:47 -05:00
cd56d774a0
parse error response for code to only change roles on 487
2013-06-27 13:00:20 -05:00
13dacdcde4
FS-5546 --resolve oddly that is the thing I was trying to fix in the first place
2013-06-26 09:17:15 -05:00
ecc4b52d4a
fix typo with some flags related to jingle
2013-06-25 14:54:34 -05:00
32adc789f6
make noexpandtab explicit in all vim modelines other than xml files
2013-06-25 11:50:17 -05:00
48e1415f64
fix return values from refactor so stun timeouts can trigger error properly
2013-06-21 12:42:37 -05:00
5a0f43d956
make sure sip-hold works on webrtc
2013-06-21 00:41:25 -05:00
a7be251bd8
fix several more windows compiler warnings and mod_gsmopen
2013-06-18 22:44:11 -05:00
ec9eebbf99
fix trivial compiler error and vs2010 reswig
2013-06-18 16:55:03 -05:00
2fac3a8e83
refactor some of the video passthru code
2013-06-17 19:52:37 -05:00
ecd8aabf5d
log auto adjust ip and port separate so it doesn't get clobbered on hold/update events
2013-05-28 16:25:39 -05:00
4ea2ae5d38
Remove extraneous parentheses
...
This resolves a clang warning:
error: equality comparison with extraneous parentheses [-Werror,-Wparentheses-equality]
2013-05-26 12:19:19 +00:00
012003b5ff
Don't setup ZRTP without a session and channel
...
FS-5472 --resolve
2013-05-25 21:10:11 +00:00
ea881b191f
media/srtp refactoring pass
2013-05-17 15:39:30 -05:00
d50366f377
FS-5418 --resolve
2013-05-15 08:08:41 -05:00
a980074f16
FS-4990 Doh! fix unintentional %s
2013-04-27 17:08:35 +08:00
5b9318c109
FS-4990 change uuid_debug_audio to uuid_debug_media and add video params
2013-04-27 06:47:06 +08:00
7a42ec06df
RTP stats were not being logged. There is no JIRA on this sigh
2013-04-26 09:31:28 -05:00
f92528436f
FS-5302 --resolve
2013-04-19 15:25:11 -05:00
4e85f9d544
fix some rtp regressions
2013-04-11 18:11:06 -05:00
455a76434c
save last sent id for validation later
2013-04-10 11:27:09 -05:00
72df7a9fe5
fix udptl issue
2013-04-09 19:43:23 -05:00
d50d43f788
make auto-adjust more logical
2013-04-08 11:00:28 -05:00
bd421d65ff
flush on answer for inbound only
2013-04-04 13:46:57 -05:00
be273b673b
windows compile another step closer
2013-04-03 15:41:46 -05:00
43eaa95746
windows compile another step closer
2013-04-03 08:55:03 -05:00
69c3c7d8e2
FS-5257 --resolve you seem to be setting sip_secure_media on an inbound leg, this is not correct. This patch should auto-correct that but also beware that the variable has changed to rtp_secure_media after this commit
2013-04-02 20:05:46 -05:00
5544db8d5e
FS-5256
2013-04-02 19:10:18 -05:00
293ed932e8
FS-5256 --resolve
2013-04-02 19:08:28 -05:00
8839d61523
fix typo
2013-04-02 13:37:04 -05:00
e334fd064f
fix variable name
2013-04-02 13:10:00 -05:00
e23b6a22bd
fix 32 bit build
2013-04-01 23:14:14 -05:00
97f0c0301f
fix zrtp build
2013-04-01 09:13:39 -05:00
fa8e199802
fix build
2013-03-31 21:27:27 -05:00
dd878da49d
clean up proxy media mode
2013-03-31 21:27:27 -05:00
45bd5952b1
fix some regressions and improve candidate selection
2013-03-31 21:27:26 -05:00
d33732ed00
MILESTONE: working firefox audio
2013-03-31 21:27:25 -05:00
c4b221c750
trying to get mozilla to work using sipml5, got stun/dtls/rtp flowing and its dead silent. c'mon guys... why do you have to do everything differently?????
2013-03-31 21:27:25 -05:00
01e411c2d7
add ice based auto-adjust
2013-03-31 21:27:25 -05:00
bc3d2b20cd
tweaks for calling local candidates
2013-03-31 21:27:24 -05:00
e1c3d51349
force the version header in the rtp packet to 2 when pulling data out of the jitter buffer because when you get a stun packet it will still be 0 from last time
2013-03-31 21:27:24 -05:00
9b16fb319a
make sure to sort rtp vs rtcp when using rtcp mux
2013-03-31 21:27:24 -05:00
6b12d7b285
fix build against older openssl
2013-03-31 21:27:23 -05:00
8d86577204
fix dtls ifdefs
2013-03-31 21:27:23 -05:00
ff532ec311
try to ifdef so we can still build against openssl older than 1.0.1 (but dtls would not work still in this case)
2013-03-31 21:27:23 -05:00
4cac512b97
change ssl symbol visibility to specific places in core only
2013-03-31 21:27:23 -05:00
16fa338b49
cleanup debug and memory and add stupid code for using sep dtls session when rtcp is not mux
2013-03-31 21:27:23 -05:00
a01ad9a1aa
dtls both ways now
2013-03-31 21:27:22 -05:00
dd9271d6b9
MILESTONE: dtls-srtp inbound
2013-03-31 21:27:22 -05:00
dabb85c3f6
wip
2013-03-31 21:27:22 -05:00
21408d4534
ice tweaks
2013-03-31 21:27:22 -05:00
b1c855e1aa
remove debug
2013-03-31 21:27:20 -05:00
d77241c414
wip
2013-03-31 21:27:19 -05:00
f1e304cc29
wip
2013-03-31 21:27:19 -05:00
aa3368a1ac
wip
2013-03-31 21:27:19 -05:00
daea865e7b
fix rtcp
2013-03-31 21:27:19 -05:00
72e2d183c1
update to cvs head srtp
2013-03-31 21:27:19 -05:00
aad4da5b71
typo
2013-03-31 21:27:18 -05:00
12e70149ff
rtcp-mux
2013-03-31 21:27:18 -05:00
1c120dd2c8
not too shabby audio and vid work both directions but browswer hates our srtcp ...
2013-03-31 21:27:18 -05:00
1db0e26e8e
wip
2013-03-31 21:27:18 -05:00
76502d9027
wip
2013-03-31 21:27:18 -05:00
aea974a91f
MILESTONE: audio and video working inbound
2013-03-31 21:27:18 -05:00
052ee0a1fa
wip
2013-03-31 21:27:17 -05:00
7510bccd6c
MILESTONE: working inbound calls
2013-03-31 21:27:17 -05:00
cb076e6b28
fucked
2013-03-31 21:27:17 -05:00
f5e2a2e0ee
fix cng
2013-03-31 21:27:17 -05:00
330f68d946
convert rtp flags to arrays
2013-03-31 21:27:13 -05:00
3226882682
FS-5203 --resolve
2013-03-27 11:09:07 -05:00
3d2457eddd
FS-5120 please update and test this version
2013-03-01 09:29:47 -06:00
b14fd4a5a4
FS-4923 try this
2013-02-14 14:07:25 -06:00
388206d784
Whitespace
2013-01-31 19:25:07 +00:00
69cda3d7cc
Fix handling of the ZRTP cache path and file name
...
libzrtp strings are weird, and we were previously misusing the
function zrtp_zstrcpyc. We can't use ZSTR_GV because it does insane
things and causes an array-bounds warning on gcc 4.7. So we have to
take matters into our own hands and setup the string correctly and
copy data into it.
Because we were doing it wrong, people would get weird pseudo-random
single-character names for the zrtp cache file, and the file would end
up in the wrong place. Now that this is fixed, users will need to
locate and move their zrtp cache file to their db_dir and name it
"zrtp.dat" if they wish to keep their current ZRTP cache.
FS-4344 --resolve
2013-01-31 18:40:24 +00:00
a46c897208
use less expensive time func in rtp stack
2013-01-15 09:15:39 -06:00
d78f4ffb19
fix vid regression
2013-01-13 21:27:17 -06:00
ca6d7a7aa0
FS-4905 This looks like it was only the case adjacent to dtmf; Try this patch.
2013-01-08 11:43:16 -06:00
6f376023a2
send cng less often
2013-01-08 11:47:15 -06:00
Michael Jerris