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572 Commits

Author SHA1 Message Date
Anthony Minessale
9e89f607c8 FS-3140 --comment-only please try this patch 2011-03-10 00:18:06 -06:00
Anthony Minessale
2a35dfb51e add rtp-notimer-during-bridge (alternative to rtp-autoflush-during-bridge 2011-03-09 15:17:26 -06:00
Anthony Minessale
8727e568e8 alter implementation of renegotiate codec on hold feature to still take other sdp elements into consideration 2011-03-08 10:37:16 -06:00
Anthony Minessale
bfd0ba9798 do not renegotiate codecs on hold re-invites 2011-03-07 13:02:41 -06:00
Anthony Minessale
89592a86e5 fix issue with polycom changing to 1 way audio on hold 2011-03-07 12:15:46 -06:00
Anthony Minessale
8fe24a2914 FS-3121 this is less of a bug and more of a feature request but here you go, that's your quota for the month 2011-03-04 12:28:41 -06:00
Mathieu Parent
316548273d Sofia: use const for variable name SWITCH_R_SDP_VARIABLE 2011-03-01 00:24:39 +01:00
Anthony Minessale
53fc3f7f78 add sip_execute_on_image variable similar to execute_on_answer etc so you can run t38_gateway or rxfax etc when you get a T.38 re-invite but no CNG tone or you want to ignore the tone and only react when getting a T.38 re-invite 2011-02-28 12:43:05 -06:00
Anthony Minessale
add9d26ac5 fix regression in video from commit c565501f555a507fa2c56eccedccdbba7a366d6d 2011-02-25 15:20:04 -06:00
Anthony Minessale
d59d41d7b4 add param to jb to try to recapture latency (disabled by default) 2011-02-25 11:59:45 -06:00
Anthony Minessale
39ff78bfae FS-3078 This is more like it 2011-02-18 20:16:11 -06:00
Anthony Minessale
25834f9537 FS-3078 NM that was a bad idea 2011-02-18 20:13:37 -06:00
Anthony Minessale
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
Anthony Minessale
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
Anthony Minessale
68d08547f3 try to improve iLBC compat 2011-02-03 16:27:22 -06:00
Anthony Minessale
74a0cfd1e1 FS-3027 2011-02-03 10:19:04 -06:00
Michael Jerris
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
Anthony Minessale
e6a25e8578 FS-2984 2011-01-14 18:42:46 -06:00
Anthony Minessale
029d68ce47 disable media timeout when encountering a recvonly stream 2011-01-14 17:42:42 -06:00
Anthony Minessale
6126383ca4 FS-2980 2011-01-13 18:41:43 -06:00
Anthony Minessale
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
Brian West
85c22d10e2 Fix iLBC when using ep_codec_string 2011-01-06 17:15:45 -06:00
Anthony Minessale
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
Anthony Minessale
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
Brian West
3734f4cd44 bump copyright date and fix some email and typos from diego. 2011-01-05 10:09:04 -06:00
Anthony Minessale
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
Anthony Minessale
668763f490 prevent race on codec change mid-call 2010-12-17 17:27:23 -06:00
Anthony Minessale
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
Anthony Minessale
7e047c3fd1 more ongoing work on jb 2010-12-14 00:15:36 -06:00
Anthony Minessale
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
Anthony Minessale
3a645dee60 FS-2913 2010-12-13 11:20:23 -06:00
Anthony Minessale
d547096164 dramatic jitterbuffer changes 2010-12-10 17:47:46 -06:00
Anthony Minessale
7aa72b67df prevent race while changing codecs mid call 2010-12-03 20:22:14 -06:00
Anthony Minessale
92f4344072 FS-2892 2010-12-01 09:46:06 -06:00
Brian West
87edbed6bb FS-535: be more careful and catch ipv6 edge case 2010-11-22 15:32:23 -06:00
Brian West
cf398e1a44 FS-535: tested but please test MORE. 2010-11-22 14:59:47 -06:00
Anthony Minessale
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
Anthony Minessale
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
Anthony Minessale
1970ec1d81 FS-2810 2010-11-01 10:03:10 -05:00
Anthony Minessale
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
Anthony Minessale
e10bc0a965 allow {dtmf_type=none} to work in oubound dial strings 2010-10-26 15:43:14 -05:00
Anthony Minessale
afaf1fac05 ilbc tweak 2010-10-15 17:53:38 -05:00
Anthony Minessale
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
Anthony Minessale
54dcb64a4d typo 2010-10-13 16:21:53 -05:00
Anthony Minessale
93c2ed941a silent recovery 2010-10-13 16:17:43 -05:00
Anthony Minessale
14361c0907 FS-620 2010-10-12 09:32:15 -05:00
Anthony Minessale
8f13eb8966 FS-2762 2010-10-06 15:17:48 -05:00
Michael Jerris
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
Anthony Minessale
589502d3d9 FS-2747 2010-10-01 14:04:06 -05:00
Anthony Minessale
56f8c11f0b refactor fmtp parser as a core func 2010-10-01 14:01:39 -05:00