b2ae5f4cc2
few bugs on recent new features
2014-10-03 15:36:23 -05:00
d52cb335db
fix trivial vs2010 build errors
2014-10-02 19:47:05 -05:00
8db31f976f
fix some recovery issues with dynamic payloads
2014-10-02 18:34:00 -05:00
10a3fa55ef
%FEATURE add bypass_media_resume_on_hold and bypass_media_after_hold variables to be set to true to enable these functions on a per channel basis
2014-10-02 17:49:09 -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
789e1481ed
FS-6880 #resolve #comment I would think that in real life once the call agreed on a codec it would only offer the negotiated codecs but we can fix this to always filter for good measure. I am not sure what the ramifications are of filtering responses but I think this patch will do so as well.
2014-10-01 13:03:50 -05:00
24084adf77
%FEATURE Add new feature to filter the SDP on bypass_media calls to remove or limit codecs.
...
VARIABLE: bypass_media_sdp_filter
Can be set globally or per leg on the inbound side of a bypass_media bridge.
VALID FILTERS:
remove(): Removes the specified codec if it exists in the SDP.
only(): Removes all codecs besides the one specified (providing that it exists in the sdp) (will not remove telephone-event))
EXAMPLE 1 (remove everything leaving only g729):
<action application="set" data="bypass_media_sdp_filter=only(g729)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 2 (remove everything leaving only g729 and also remove dtmf):
<action application="set" data="bypass_media_sdp_filter=only(g729)|remove(telephone-event)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 3 (remove alaw and speex):
<action application="set" data="bypass_media_sdp_filter=remove(pcma)|remove(speex)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
2014-10-01 01:28:10 +05:00
fbe857e6fa
fix ptime from known broken endpoints on re-invite
...
Freeswitch tries to fix timing issues (wrong ptime) on re-invite the same way
it does for the initial invite. This results in small audio glitches, while it
sends a couple of packets with different ptime, before the timing detection
logic figures out the remote (broken) endpoint true ptime.
In order to avoid unnecessary timing changes, this patch overwrites the
advertised ptime from known broken endpoints with the ptime, which was detected
by freeswitch. It does this by checking if the sip_h_X-Broken-PTIME (1.2.x) or
rtp_h_X-Broken-PTIME (master) variables are set.
FS-6644 #resolve
2014-09-30 11:19:35 +02:00
372455c30a
FS-6829 #resolve
2014-09-19 02:28:47 +05:00
d2f8fca18a
FS-6825 #resolve #comment caused by regression in commit 0732c0b0 pertaining to FS-6825
2014-09-17 20:32:18 +05:00
9215c93827
FS-6785 - pass and use sample rate in switch_core_session_get_payload_code()
2014-09-08 14:26:58 +00:00
5173471b06
nm
2014-08-16 02:00:38 +05:00
cb928cdf37
fix typo
2014-08-16 01:52:03 +05:00
b2162142f3
FS-6667 #comment block sending audio when dmachine is reading
2014-08-06 01:21:34 +05:00
f629b5ed63
always start payloads at 102 to avoid webrtc funk
2014-08-02 00:59:44 +05:00
502ddd312a
FS-6702 #resolve
2014-07-29 22:17:02 +05:00
0732c0b027
FS-6631 #resolve
2014-07-29 03:04:29 +05:00
e7700c0ac4
FS-6665 #resolve #comment This should fix it
2014-07-18 22:05:15 +05:00
6dca24bb2a
Merge pull request #5 in FS/freeswitch from ~KATHLEEN.KING/freeswitch-fork:clang-warnings-werror to master
...
* commit '1affff9db4aaa29ab66f9f5db76f575eeabd86b5':
Fixed a clang-3.5 missing-prototype warning and added doxygen documentation for switch_channel_set_presence_data_vals. #doxygen
Fixed dead code.
Fixed trucation of value warning.
Removed a useless called to abs.
Removed an autological-pointer-compare from src/switch_utils.c.
Fixed trucation of value warning.
2014-07-14 17:38:21 -05:00
831d5892b9
preserve timer name for recovery
2014-07-03 23:50:14 +05:00
89f7196a6e
Fixed trucation of value warning.
...
There was a parameter mismatch between abs(), which expects an int,
and atol() which returns a long. Since max_drift is defined as an int,
there is no need to pars q as a long rather than an int.
2014-07-03 10:53:19 -07:00
b9a51bbf91
FS-6632 Trying to actually use opus in stereo broke pretending to use it in stereo
2014-06-25 21:19:07 +05:00
faeb00368e
verto bug fixing in codec negotiation and some stereo codec stuff
2014-06-20 01:41:28 +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
4ffbf23ffa
stereo refactoring
2014-06-13 01:49:16 -04:00
0d7b1ff42a
chrome now requires longer ice password
2014-05-29 15:46:31 -04:00
dcf633b1dd
Revert workaround of gcc-4.6.3-14+rpi1 ARM bug
...
The correct fix is to use gcc-4.7 or newer. If the compiler borked
this, one should trust nothing else it produces.
Revert "FS-6524,FS-6494: the compiler for rpi seems to be strangely
mis-optimizing this if/else block. Yes, its already set to this value,
but without the else block, it 'magically' hits the if blocks above,
unless you sleep, print a log, or add the else"
This reverts commit 1ca1d63795
2014-05-22 01:03:24 +00:00
bedf8327b6
FS-6418 FS-6427 add profile param NDLB-proxy-never-patch-reinvites to solve both issues
2014-05-21 03:31:50 +05:00
1ca1d63795
FS-6524,FS-6494: the compiler for rpi seems to be strangely mis-optimizing this if/else block. Yes, its already set to this value, but without the else block, it 'magically' hits the if blocks above, unless you sleep, print a log, or add the else
2014-05-20 16:20:36 -05:00
4ec5e5af47
Refactor if blocks
...
Since the first branch doesn't need to do anything, we invert it and
pull a sub-branch out a level into the main conditional.
2014-04-30 23:23:06 +00:00
ae3cf0a1a9
Remove dead assignment
...
...now this code needs refactoring.
2014-04-30 23:22:13 +00:00
d0de8c0bc6
add heartbeat_use_scheduler variable to force the heartbeat to use the scheduler rather than the media stream and also make it auto-chage between modes when going in and out of the media path
2014-04-29 04:20:51 +05:00
de7d5a9e07
Resolve key length issue with AES GCM 128/256
2014-04-08 11:04:41 -05:00
72c52052dc
FS-6436
2014-04-08 20:30:23 +05:00
340b697e1b
FS-6341: --resolve add 3pcc invite w/o sdp support for 100rel/PRACK
2014-03-17 12:27:42 -04:00
0fa6cc6f01
FS-6340 --resolve
2014-03-11 11:38:31 -05:00
8d2c6b354e
According to https://code.google.com/p/webrtc/issues/detail?id=2768 ; The Chrome WebRTC engine reserves payload 98 and 99, IKR? So, to avoid taking a nasty spill down the stairs and subjecting ourselves to further school absences, we'll just start our payload space at 102 when making WebRTC calls.......
2014-03-07 20:24:16 +05:00
b8e4a66dd2
another crypto regression
2014-03-07 08:34:39 +05:00
5aa955b5c9
also means forbidden on recovering calls
2014-03-07 07:11:32 +05:00
87e0dda3d3
no var set on outbound meands forbidden and on inbound it means optional
2014-03-06 20:05:02 -06:00
fcef3ad4b1
FS-6319 --resolve
2014-03-06 19:37:11 -06:00
46c5268e09
FS-6319
2014-03-07 06:35:02 +05:00
bd4a0d8cbc
add a way to tell mod_conference when the rate of the channel has changed due to a codec change so it can reset the resampler and codecs internally
2014-03-07 05:17:47 +05:00
f7be96396b
add missing flags after refactor
2014-03-06 10:49:43 -06:00
656cb2ac03
add optional rtp_secure_media_suites variable clobbered by rtp_secure_media with mandatory|optional:<suites>
2014-03-06 08:17:22 +05:00
1d733235a5
remove unused stuff from last commit
2014-03-06 07:41:39 +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
d3121d930e
switch_false currently returns false on NULL
2014-03-05 17:36:35 -06:00
6ae038add3
FS-5755 84c0680153 is wrong. It made optional enforce crypto.
2014-03-04 19:07:54 -06:00
84c0680153
FS-5755 Fix regression if rtp_secure_media=false, it will force encryption.
2014-03-04 09:42:17 -05:00
15f4bd44ca
FS-5886 --resolve add NDLB_broken_opus_sdp which must be set as a global variable in var.xml or switch.conf.xml to enable broken opus mode for back compat with counterpath mobile
2014-02-28 21:57:47 +05:00
3dad15f938
FS-5755 part 2
...
rtp_secure_media=true
--inbound: Accept the srongest supported offered crypto suite, MUST result in a negotiated crypto or aborts.
--outbound: offer all supported crypto suites, MUST result in a negotiated crypto or aborts.
rtp_secure_media=optional
--inbound: Accept the srongest supported offered crypto suite, fall back to no crypto if no valid ones accepted.
--outbound: offer all supported crypto suites, OPTIONAL result in a negotiated crypto falls back to no crypto.
rtp_secure_media=<suite1>,<suiteN>
--inbound: same behaviour as rtp_secure_media=true with smaller set of acceptable suites.
--outbound: offer supplied crypto suites, same behaviour as rtp_secure_media=true with smaller set of suites.
2014-02-28 02:10:56 +05:00
3e499da329
FS-6275 --resolve I don't think this is a problem with latest firefox + latest master but i'll ensure it with this change. Anyone into testing webrtc should always use master until an official 1.4 is released.
2014-02-26 21:35:38 +05:00
5646957c5b
FS-5937
2014-02-26 04:06:59 +05:00
33780fca07
FS-5937 now with more working
2014-02-25 09:55:21 -06:00
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
b41d3a2c4a
FS-5755 minor regression
2014-02-25 03:15:48 +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
c945584201
FS-6237 --resolve
2014-02-17 23:48:59 +05:00
b4b02f1eda
FS-6002 -master
2014-02-17 22:25:19 +05:00
63c415ac38
FS-6229 try this
2014-02-17 22:07:38 +05:00
6e7d5d0897
update copyright header for 2014
2014-02-12 12:08:56 -06:00
a0181631d1
don't fail when rtp_write returns 0
2014-02-12 08:29:09 +05:00
3ecb504fda
FS-6186 --resolve
2014-02-05 07:15:08 +05:00
0148389c82
FS-6170 please test
2014-01-31 00:26:36 +05:00
6bec36168c
FS-6153 try this revision
2014-01-30 23:02:28 +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
7104ecac28
FS-6153 try this patch
2014-01-30 02:51:07 +05:00
dfb3a4becc
FS-6094 --resolve amendment
2014-01-29 02:19:03 +05:00
358b5db0e4
block all possible routes to write data during ice and dtls negotiation
2014-01-25 03:48:32 +05:00
a71879262a
FS-6094 --resolve This change should fix the problem and you don't need to try the patch anymore
2014-01-25 00:18:10 +05:00
bd7182c390
FS-5755 part 1
2014-01-24 05:44:36 +05:00
8dfbd91a3d
fold version macros into runtime functions to avoid cascade rebuilds when modifying the core
2014-01-17 01:54:47 +05:00
1b9c21ee88
some webrtc tweaks
2014-01-15 07:04:24 +05:00
6c712045fd
FS-6108 --resolve caused by switch_ivr_media while on hold
2014-01-14 22:59:49 +05:00
3ca06e1ea7
FS-6073 --resolve
2014-01-03 23:45:35 +05:00
0ab8cf0f7c
FS-6054
2013-12-19 21:36:28 +05:00
417ef16de5
FS-6054
2013-12-19 21:32:44 +05:00
9a20d0ab0c
FS-6054 --resolve
2013-12-18 23:22:29 +05:00
8231cb9f01
fix codec renegotiation on hold/reinvite issue
2013-12-16 23:46:29 +05:00
2b60308891
redo
2013-12-01 02:53:27 -06:00
2f7ed973ed
separate read mutex for each stream type
2013-11-30 20:49:59 -06:00
ae99177598
add more range to saved candidates and fix off-by-one error when total is met
2013-11-30 19:13:49 -06:00
c56b73d0be
remove unused variable
2013-11-26 17:15:14 +00:00
83ccceb2b5
FS-5958 remove debug
2013-11-26 10:04:29 -06:00
2747660269
FS-5959 FS-5958
2013-11-25 12:08:17 -06:00
3a0f934788
revert one change
2013-11-22 18:52:23 +05:00
b8dc07132a
FS-5959 i hate this ticket
2013-11-22 08:55:45 +05:00
ccaa3ae732
FS-5959
2013-11-21 01:38:21 +05:00
51464b7b8c
FS-5959 always do rtp_mirror_fmtp behaviour
2013-11-20 23:49:18 +05:00
8c03520502
FS-5959
2013-11-20 22:40:47 +05:00
0030ce440e
FS-5958 this should pass it through in late mode too
2013-11-20 05:20:01 +05:00
ce690418a1
FS-5959 add rtp_mirror_fmtp var
2013-11-20 03:36:51 +05:00
714eb245fd
remove unused code
2013-11-19 15:19:12 -06:00
f14260e8dd
set fmtp in payload_map init function
2013-11-19 00:43:30 +05:00
7330b005da
codec tweak
2013-11-15 01:17:22 +05:00
6e372418d9
fix seg in new media code
2013-11-14 22:19:10 +05:00
98d453c7d9
move media reset code a little deeper so it does not interrupt media when there is no change
2013-11-14 04:18:59 +05:00
043376b2db
tweak
2013-11-12 22:17:58 +05:00
0d1b647197
remainder of previous patch to pass through dynamic payload numbers used
2013-11-12 00:42:32 +05:00
13dcbae0b4
regression from last commit
2013-11-10 22:11:37 +05:00
fb78273ab0
inherit dynamic payloads from A leg where applicable
2013-11-10 07:53:48 +05:00
a4dd6ccb42
fix some issues in passthrough of fec packets
2013-11-09 23:02:11 +05:00
bf03df9f14
fix windows compiler warnings and vs2010 reswig
2013-11-07 23:01:40 -06:00
bb9adff511
rework media code
2013-11-08 03:48:00 +05:00
a7dab0464d
FS-5933 --resolve
2013-10-31 11:37:46 -05:00
30f7480c9c
FS-5680 revert, set this variable globally in vars.xml instead
2013-10-30 14:21:21 -05:00
abc230ed50
FS-5909 use correct ip in patched sdp
2013-10-29 12:58:53 -05:00
8244fb62e8
more opus neg drama
2013-10-29 03:44:24 +05:00
ce83817318
webrtc tweaks
2013-10-28 23:39:33 +05:00
6b2c9e4478
change codec negotiation for edge opus cases
2013-10-24 01:23:18 -05:00
9246309a21
comment out
2013-10-17 12:07:37 -04:00
b694cb09a6
google changed something in DTLS
2013-10-17 03:58:09 +05:00
4a172402d2
some refactoring and plumbing for 1.4
2013-10-16 02:24:32 +05:00
f613105a25
FS-4865 --resolve
2013-09-16 22:23:49 +05:00
3392db98bf
FS-5781 --resolve try now
2013-09-12 20:39:57 +05:00
98a092b37d
FS-5766 --resolve
2013-09-10 03:03:10 +05:00
b513d1ff02
fix possible issue with double read of frames in conf
2013-09-06 00:05:00 +05:00
0dcbc18322
fix issue with default video echo
2013-09-04 02:58:05 +05:00
d35db852a8
follow up to last commit
2013-08-16 12:17:00 -04:00
80e2633bb7
add mutex in media_engine to prevent double read in video thread
2013-08-16 20:34:34 +05:00
ef62351c47
FS-5693 --resolve
2013-08-14 02:00:36 +05:00
3682e54d19
FS-5680 --resolve this patch should auto set this variable on dtls enabled channels
2013-08-08 04:20:36 +05:00
39ad799683
FS-4932 FS-5090 --resolve
2013-07-29 10:18:05 -05:00
a4bc98a775
FS-5643 --resolve this is the kind of regressions we need to keep an eye on from the refactoring of the media code. I removed the codec_ms from the engine in addition to applying your patch
2013-07-26 10:24:19 -05:00
84c3e3ade2
FS-5572 --resolve set new variable by doing export bypass_keep_codec=true to force re-invite with the same codec it was using previously
2013-07-25 14:12:26 -05:00
3ddbad4dd3
let sofia recover work in some tcp situations
2013-07-23 13:52:49 -05:00
eebac2ea95
webrtc video tweaks
2013-07-16 19:16:01 -05:00
eff5f3a766
add debug
2013-07-16 09:05:06 -05:00
82ffd425c7
cleanup
2013-07-12 14:27:10 -05:00
5dbf2b3cf7
refactor some video code
2013-07-11 17:38:24 -05:00
ba96b391d3
fix trivial compiler error
2013-07-02 12:41:15 -05:00
134871872c
fix codec negotiation issue with the same codec at multiple rates
2013-07-02 12:16:54 -05:00
bf5fa172e5
FS-5547 --resolve
2013-06-26 10:47:40 -05:00
32adc789f6
make noexpandtab explicit in all vim modelines other than xml files
2013-06-25 11:50:17 -05:00
5a0f43d956
make sure sip-hold works on webrtc
2013-06-21 00:41:25 -05:00
9cc21d28fc
add novideo app
2013-06-20 13:10:21 -05:00
4baf0276d6
FS-5513 --resolve
2013-06-18 21:16:56 -05:00
d642eb96f0
FS-5513 --resolve
2013-06-17 21:37:53 -05:00
2fac3a8e83
refactor some of the video passthru code
2013-06-17 19:52:37 -05:00
01959606cd
FS-5493 oops
2013-06-06 13:17:20 -05:00
75fab9b0f2
FS-5493 --resolve
2013-06-06 12:59:18 -05:00
bc25079fce
Fix spelling in log message
2013-06-02 20:06:34 +00:00
699bd795f6
FS-5455 try this
2013-05-22 08:11:29 -05:00
2bafe135c0
remove flag
2013-05-17 16:13:19 -05:00
ea881b191f
media/srtp refactoring pass
2013-05-17 15:39:30 -05:00
644720454f
fix some minor regression in media code
2013-05-14 12:55:02 -05:00
fd86c038c9
FS-5406 --resolve
2013-05-12 08:06:31 -05:00
ba97929a1a
FS-5398 --resolve
2013-05-09 10:08:14 -05:00
33f08490ec
refuse t38 passthru on unanswered channel
2013-05-02 22:29:02 -05:00
5b9318c109
FS-4990 change uuid_debug_audio to uuid_debug_media and add video params
2013-04-27 06:47:06 +08:00
8ab04152f6
FS-4990
2013-04-26 11:07:50 -05:00
7a42ec06df
RTP stats were not being logged. There is no JIRA on this sigh
2013-04-26 09:31:28 -05:00
de1f013f2e
FS-5331
2013-04-23 18:08:41 -05:00
25de1531ce
FS-5331
2013-04-23 17:59:26 -05:00
9bf19ec74d
FS-5331
2013-04-23 14:27:20 -05:00
5e71ab6a06
FS-5331 --resolve yes this is a bug that is now fixed
2013-04-23 10:13:54 -05:00
b1447dd635
fix codec issue on recover
2013-04-16 13:02:59 -05:00
fda6e896e2
try to be back compat with the more logical yet incorrect sdp format for opus
2013-04-16 09:25:02 -05:00
ae95109eb7
fix srtp neg in rare case
2013-04-08 13:24:39 -05:00
fd7d5e0801
windows compiler fixes - now builds again
2013-04-03 21:55:50 -05:00
be273b673b
windows compile another step closer
2013-04-03 15:41:46 -05:00
faea5f0c08
FS-5258 --resolve
2013-04-03 09:02:13 -05:00
de6389dd4f
workaround for silly https://code.google.com/p/chromium/issues/detail?id=177099
2013-03-31 21:27:27 -05:00
dd878da49d
clean up proxy media mode
2013-03-31 21:27:27 -05:00
157a0b8a1e
fix seg in multiple m line
2013-03-31 21:27:27 -05:00
ee1b4db480
fix FS-5116 the new way
2013-03-31 21:27:26 -05:00
3201478830
sigh
2013-03-31 21:27:26 -05:00
45b4ba1afa
add webrtc_enable_dtls=false (for webrtc) or enable_dtls=true (for normal calls)
2013-03-31 21:27:26 -05:00
45bd5952b1
fix some regressions and improve candidate selection
2013-03-31 21:27:26 -05:00
b93fc016b8
move verbose_sdp to a flag and set it to true in webrtc calls
2013-03-31 21:27:25 -05:00
4b3aa396fb
fix some regessions and sync namespace of some vars
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
05dd02d0a6
make opus work, omfg .. serious??? you need to always advertise it as stereo using the standard way then specify it in the fmtp insread just to complicate things.... that's not cool =/
2013-03-31 21:27:25 -05:00
190b31871d
fix auto-select of wan.auto when no candidates defined
2013-03-31 21:27:25 -05:00
1aecb1a9c7
remove wtf
2013-03-31 21:27:24 -05:00
bc3d2b20cd
tweaks for calling local candidates
2013-03-31 21:27:24 -05:00
2aade020e9
fix dtmf and cng neg
2013-03-31 21:27:24 -05:00
4711fd2b92
same issue
2013-03-31 21:27:24 -05:00
53457e05bf
block code when no dtls is present
2013-03-31 21:27:24 -05:00
2b6a0ce0a4
don't respond w/ dtls when we don't support it
2013-03-31 21:27:23 -05:00
a6a055cabb
fix or and thing
2013-03-31 21:27:23 -05:00
32bc398081
remove printf
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
ad1226b5c4
change name to match steves changes in master
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
9cc550ee18
build sdp lib inline w/ core
2013-03-31 21:27:22 -05:00
668f4a6207
working mo better
2013-03-31 21:27:20 -05:00
5c90322e8c
MILESTONE: kind of works on media and signaling
2013-03-31 21:27:20 -05:00
b7eac54be5
only send srflx when we need one
2013-03-31 21:27:20 -05:00
aba03f9e6c
set controlling when appropriate
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
12e70149ff
rtcp-mux
2013-03-31 21:27:18 -05:00
36d1388a99
forgot
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
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
20bbee8765
fix sdp gen
2013-03-31 21:27:17 -05:00
6939394530
isolate sdp parser code to the switch_core_media.c
2013-03-31 21:27:17 -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
5490cdf456
wip
2013-03-31 21:27:17 -05:00
20568e64df
wip
2013-03-31 21:27:17 -05:00
f5e2a2e0ee
fix cng
2013-03-31 21:27:17 -05:00
241300fb56
port over revovery code
2013-03-31 21:27:16 -05:00
078d963222
MILESTONE: all the crap out of tech_pvt and sofia in general related to media shit
2013-03-31 21:27:16 -05:00
85308861d8
almost rid of crufty tech_pvt members
2013-03-31 21:27:16 -05:00
75c95fa98e
MILESTONE: working audio and no reference to rtp_session in all of sofia
2013-03-31 21:27:15 -05:00
92ddf47293
expose some rtp func via media handle
2013-03-31 21:27:15 -05:00
d012c5f392
omg it can make calls ... long way to go ...
2013-03-31 21:27:15 -05:00
48554d1a85
compiles
2013-03-31 21:27:15 -05:00
3c51053bb4
compiles the core wow
2013-03-31 21:27:14 -05:00
eb81042a6f
wip
2013-03-31 21:27:14 -05:00
d6af3a1fdd
omfg
2013-03-31 21:27:14 -05:00
10010d47c1
wip
2013-03-31 21:27:14 -05:00
e747229f5e
typo
2013-03-31 21:27:13 -05:00
0a246d1cb4
pull srtp negotiation stuff into the core stage 1
2013-03-31 21:27:13 -05:00
330f68d946
convert rtp flags to arrays
2013-03-31 21:27:13 -05:00
592993ecda
baseline
2013-03-31 21:27:13 -05:00
Anthony Minessale