e70af1f84f
don't nat map on loopback addrs
2011-04-29 10:28:56 -05:00
73279f01bf
FS-3166 --resolve
2011-04-22 16:43:29 -05:00
5857495e06
offer both avp and savp when using srtp
2011-04-15 11:17:36 -05:00
ceed7658e2
check_decode the caller id name in this case too
2011-04-14 19:09:11 -04:00
4b706dac51
FS-3227 --resolve this looks like sane changes. My only complaint was the formatting. Watch for whitespace indentation by looking at the code in emacs or vi where it should be tabed properly.
2011-04-04 11:55:05 -05:00
fda2283bbd
auto-aleg-full and auto-aleg-domain for from_domain field in gateway
2011-04-03 12:03:29 -05:00
8312d74121
FS-2819 --comment-only please try this patch
2011-03-30 11:26:19 -05:00
81bfe43589
mod_sofia: Correct a problem where restarting profile would cause some profile hash entry to remain.
2011-03-25 15:50:52 -04:00
e657e32fca
FS-3172 this also fixes the incorrect usage of L16 on payload 10 which may or may not break interop with other sip devices if we do it right. also added rtp_disable_byteswap variable that can be set to false to disable byteswap when a device is encountered that is incompat (inluding all precious version of FS up till now)
2011-03-21 14:31:16 -05:00
db7933e72b
jitter buffer sanity checks
2011-03-17 22:29:16 -05:00
4832d26a3a
put this back to 0
2011-03-10 15:32:09 -06:00
9e89f607c8
FS-3140 --comment-only please try this patch
2011-03-10 00:18:06 -06:00
2a35dfb51e
add rtp-notimer-during-bridge (alternative to rtp-autoflush-during-bridge
2011-03-09 15:17:26 -06:00
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
bfd0ba9798
do not renegotiate codecs on hold re-invites
2011-03-07 13:02:41 -06:00
89592a86e5
fix issue with polycom changing to 1 way audio on hold
2011-03-07 12:15:46 -06:00
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
316548273d
Sofia: use const for variable name SWITCH_R_SDP_VARIABLE
2011-03-01 00:24:39 +01:00
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
add9d26ac5
fix regression in video from commit c565501f55
2011-02-25 15:20:04 -06:00
d59d41d7b4
add param to jb to try to recapture latency (disabled by default)
2011-02-25 11:59:45 -06:00
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
Anthony Minessale