39ff78bfae
FS-3078 This is more like it
2011-02-18 20:16:11 -06:00
25834f9537
FS-3078 NM that was a bad idea
2011-02-18 20:13:37 -06:00
a23b335b50
FS-3078 see wrapper function that should do the same thing this is called at the time when the sdp is created so if it still doesn't work it would suggest that you have this variable set passing in from the other leg in which case you need to set it explicitly because the mode of the inbound leg prevails over the profile default
2011-02-18 19:03:07 -06:00
c565501f55
tell rtp stack about what remote payload type to expect when the receiving end follows the stupid SHOULD as WONT and sends a different dynamic payload number than the one in the offer
2011-02-15 16:09:58 -06:00
68d08547f3
try to improve iLBC compat
2011-02-03 16:27:22 -06:00
74a0cfd1e1
FS-3027
2011-02-03 10:19:04 -06:00
018a3800b4
fix session timer failure when freeswitch is generating the sdp and there are enough dynamic codecs enabled to conflict with the 2833 pt (4 by default)
2011-01-17 13:11:10 -06:00
e6a25e8578
FS-2984
2011-01-14 18:42:46 -06:00
029d68ce47
disable media timeout when encountering a recvonly stream
2011-01-14 17:42:42 -06:00
6126383ca4
FS-2980
2011-01-13 18:41:43 -06:00
b3fc001e6c
add rtp_bug IGNORE_DTMF_DURATION to speed up dtmf detection of RFC2833 on strange carriers
2011-01-07 16:04:24 -06:00
85c22d10e2
Fix iLBC when using ep_codec_string
2011-01-06 17:15:45 -06:00
b262f44ce2
add temp_hold_music var that is only valid until you transfer the call and finishing touches on bind meta to A-D
2011-01-05 18:58:56 -06:00
181b543b0c
add auto-jitterbuffer-msec param and auto-disable the jitterbuffer when briding to another channel who also has a jitterbuffer so both legs will disable during a bridge
2011-01-05 16:25:14 -06:00
3734f4cd44
bump copyright date and fix some email and typos from diego.
2011-01-05 10:09:04 -06:00
97a68c50d9
support allowing pidf-ful presence clients to share the same account and 'appear offline' without influencing each other =/ also refactor the contact generation string based on nat into a helper function
2010-12-30 11:38:23 -06:00
668763f490
prevent race on codec change mid-call
2010-12-17 17:27:23 -06:00
93cc3dc556
normalize tests for outbound channels to use switch_channel_direction instead of testing for CF_OUTBOUND
2010-12-15 20:59:42 -06:00
7e047c3fd1
more ongoing work on jb
2010-12-14 00:15:36 -06:00
321013efe7
have mod_sofia always elect to be the session refresher so we know it will work, also make the session-expires set to 0 imply 100% disabled session timers
2010-12-13 14:02:46 -06:00
3a645dee60
FS-2913
2010-12-13 11:20:23 -06:00
d547096164
dramatic jitterbuffer changes
2010-12-10 17:47:46 -06:00
7aa72b67df
prevent race while changing codecs mid call
2010-12-03 20:22:14 -06:00
92f4344072
FS-2892
2010-12-01 09:46:06 -06:00
87edbed6bb
FS-535: be more careful and catch ipv6 edge case
2010-11-22 15:32:23 -06:00
cf398e1a44
FS-535: tested but please test MORE.
2010-11-22 14:59:47 -06:00
6c4f49a888
apparently some sip device vendors did not read the RFC (who knew?) adding verbose_sdp=true var to add needless a= lines for standard iana codecs that explicitly do not require them
2010-11-19 13:46:14 -06:00
b278dd2379
add manual_rtp_bugs to profile and chan var and 3 new RTP bugs SEND_LINEAR_TIMESTAMPS|START_SEQ_AT_ZERO|NEVER_SEND_MARKER
...
RTP_BUG_SEND_LINEAR_TIMESTAMPS = (1 << 3),
Our friends at Sonus get real mad when the timestamps are not in perfect sequence even during periods of silence.
With this flag, we will only increment the timestamp when write packets even if they are eons apart.
RTP_BUG_START_SEQ_AT_ZERO = (1 << 4),
Our friends at Sonus also get real mad if the sequence number does not start at 0.
Typically, we set this to a random starting value for your saftey.
This is a security risk you take upon yourself when you enable this flag.
RTP_BUG_NEVER_SEND_MARKER = (1 << 5),
Our friends at Sonus are on a roll, They also get easily dumbfounded by marker bits.
This flag will never send any. Sheesh....
2010-11-10 16:58:36 -06:00
1970ec1d81
FS-2810
2010-11-01 10:03:10 -05:00
19325c4369
fix race in codec failure condition, then fix bug in sdp parsing (likely a regression from recent codec changes) to never have the problem in the first place so you are double-protected
2010-10-27 16:37:35 -05:00
e10bc0a965
allow {dtmf_type=none} to work in oubound dial strings
2010-10-26 15:43:14 -05:00
afaf1fac05
ilbc tweak
2010-10-15 17:53:38 -05:00
dfa78985b4
Change codec behaviour
...
channel_variable: sdp_m_per_ptime
Adds a new m= line for each distinct ptime in codec list.
When this variable is not set:
When mixing codecs with various ptime in a codec list, they will now be allowed to co-exist in the sdp but it will send no ptime attr.
This means the ptime preferences on the offer will be ignored when mixing codecs with various ptimes.
When receiving a codec list with no ptime attr, the ptime will be chosen from local preference instead of assuming 20ms
This means if offer contains PCMU with not ptime and FS has PCMU@40i
Dynamic payloads will now start at 98 and increment per additional dynamic codec per call.
So now you can add CELT@32000h,CELT@48000h and each one will be auto-assigned a dynamic paylaod type.
2010-10-13 19:28:20 -05:00
54dcb64a4d
typo
2010-10-13 16:21:53 -05:00
93c2ed941a
silent recovery
2010-10-13 16:17:43 -05:00
14361c0907
FS-620
2010-10-12 09:32:15 -05:00
8f13eb8966
FS-2762
2010-10-06 15:17:48 -05:00
c701d41c3c
add sofia_glue_find_parameter_value function to get a specific value from a url params string
2010-10-03 20:00:32 -04:00
589502d3d9
FS-2747
2010-10-01 14:04:06 -05:00
56f8c11f0b
refactor fmtp parser as a core func
2010-10-01 14:01:39 -05:00
f8c9ef5f9d
fix codec regression from yesterday
2010-09-30 15:15:58 -05:00
03e74c51f9
FS-526: improve nat handling when using stun or host as ext-rtp-ip
2010-09-30 10:23:48 -05:00
633f193dae
add bitrate patch from moc with some extra stuff for late neg mode
2010-09-29 16:52:43 -05:00
45ecbc2f61
fix warnings on windows x64 builds src and mods projects - only libsofia included on the libs side
2010-09-29 15:47:58 -05:00
84a383fe0f
improve video support for new polycom phones
2010-09-29 14:14:41 -05:00
45e2b99dab
don't passthru when its proxy media, bypass media or there is no rtp session, fixes seg
2010-09-27 13:11:51 -05:00
ed708bff3e
update
2010-09-24 15:14:58 -05:00
73e6e03d37
MODSOFIA-105 can't take defaults that were there before away
2010-09-24 11:29:06 -05:00
8f731f42e2
MODSOFIA-105
2010-09-24 10:31:11 -05:00
a9446ac12e
SFSIP-227
2010-09-23 11:25:43 -05:00
Anthony Minessale