Simon Morlat
31d767f9e3
fix compilation error
2014-06-04 15:17:48 +02:00
Simon Morlat
10c9de93ca
implement early media forking at client side
2014-06-04 15:16:21 +02:00
Jehan Monnier
ba5c902bba
add option sip_update to linphonerc to disable SIP UPDATE
2014-06-03 10:24:44 +02:00
Simon Morlat
a5af301c13
fix memory leaks
2014-06-02 17:33:34 +02:00
Ghislain MARY
0cd71d6548
Fix issue with rtcp-fb attributes in SDP of response.
2014-06-02 11:02:41 +02:00
Ghislain MARY
ad64b94401
Parse rtcp-fb attributes contained in SDP.
2014-06-02 11:02:41 +02:00
Ghislain MARY
0a40048d4b
Add rtcp-fb attributes in the SDP.
2014-06-02 11:02:41 +02:00
Ghislain MARY
2110281d2e
Handle AVPF and SAVPF profiles.
2014-06-02 11:02:40 +02:00
Simon Morlat
fbc8f77e3a
allow crypto lines to be configured from linphonerc, and improve code handling SRTP crypto lines
2014-05-21 13:11:13 +02:00
Simon Morlat
a63a3f59cd
avoid sending dummy route headers as much as possible
2014-05-09 15:20:14 +02:00
Jehan Monnier
b5117cf71e
Answer 504 to UPDADE requests received in a confirmed dialog
2014-05-05 12:03:46 +02:00
Simon Morlat
e942c6590b
fix clang warning
2014-05-05 11:03:46 +02:00
Simon Morlat
4296c3945c
update oRTP, fix bad error output, and restore UPDATE method in allow header (removed by mistake)
2014-05-02 23:22:36 +02:00
Simon Morlat
3a6aa9f08d
deep modifications about audio & video codec bitrates are handled.
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- vbr codecs can automatically have different output bitrates depending on whether video is used and/or allowed total output bandwidth
- application can specify an output IP bitrate for a given codec, which allows to control the quality of vbr codecs.
Note: a belle-sip upgrade is required to fix a bug around channels parsing in rtpmap.
2014-05-02 20:24:51 +02:00
Simon Morlat
f6d63524d3
fix declared number of channels in SDP for opus codec, to follow opus-rtp draft.
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add ugly hack to allow older versions of linphone to call new versions with opus.
2014-05-01 12:14:05 +02:00
Simon Morlat
7553aa6492
- linphone now puts Route headers in requests (except register) for outbound proxy configurations, according to RFC3261
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This behavior can be reverted by putting [sip]->use_no_initial_route=1 in the configuration file.
- accept presence NOTIFY without bodies, instead of replying 415
- remove belle-sip warning at start due to stack not created early enough.
2014-04-25 23:13:26 +02:00
Simon Morlat
811223d35f
reset error_info in case of success
2014-04-23 16:57:10 +02:00
Gautier Pelloux-Prayer
4386f18b21
replace tabs with spaces and remove trailing spaces
2014-04-22 17:22:51 +02:00
Jehan Monnier
43aa6ef34f
change encryption state management
2014-04-17 16:22:49 +02:00
Simon Morlat
39f9ec6a48
improve LinphoneEvent api:
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- better error notification
- allow publish without expires
2014-04-11 10:00:13 +02:00
Simon Morlat
7aec150bf4
fix missing custom header processing in generic PUBLISH api
2014-04-10 19:43:16 +02:00
Simon Morlat
bd83f0b7ca
fix crash while receiving a SIP message without content-type.
2014-04-09 00:00:22 +02:00
Simon Morlat
a8176a398d
rework SRTP support so that recv and send key can be set and updated independently.
2014-04-08 23:41:14 +02:00
Jehan Monnier
9b4197fef9
add retry algo for vfu request + enable opus cbr by default
2014-04-05 09:45:06 +02:00
Simon Morlat
274d50168e
implement digest authentication for anonymous calls (with id privacy)
2014-04-02 22:23:52 +02:00
Simon Morlat
3013fd8ae2
allow configuration of root_ca before provisioning, so that it can be used for https fetching
2014-03-26 17:51:47 +01:00
Ghislain MARY
1c3714327f
Do not use raw attributes to get values.
2014-03-26 16:58:13 +01:00
Simon Morlat
266207c5f0
change behavior of linphone_core_get_sip_transports() if random port selection was specified.
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Only linphone_core_get_sip_transports_used() will return the real port if random port selection was specified.
2014-03-26 11:30:07 +01:00
Simon Morlat
250495034e
rely on belle-sip ability to choose SIP transport random port using bind().
2014-03-25 22:48:17 +01:00
Simon Morlat
c25273e9ca
add C function to disable chat
2014-03-25 12:48:50 +01:00
Simon Morlat
269f8d1c4e
add new API to obtain full details about failures (calls, registration, events).
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Fix bug when receiving a 487 after cancelling call, resulting in a call waiting tone to be played.
2014-03-21 18:15:28 +01:00
Ghislain MARY
681b445470
Prevent duplication of RTCP XR attribute in the medias when identical to the session one.
2014-03-19 16:07:05 +01:00
Ghislain MARY
9a4fd36948
Configure the RTCP XR parameters of the rtp session.
2014-03-19 16:07:04 +01:00
Ghislain MARY
4c3baa0528
Define RTCP XR structure in oRTP instead of sal.
2014-03-19 16:07:04 +01:00
Jehan Monnier
260e7e1d5c
minimal SIP UPDATE support
2014-03-18 09:08:25 +01:00
Simon Morlat
c7f23e6494
set default value of video renderer
2014-03-17 11:40:36 +01:00
Simon Morlat
a0b7b1f3b0
rework implementation of SUBSCRIBEs delayed to successful registration.
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In case of network errors, SUBSCRIBE could be sent before registration refresh, this should be fixed.
2014-03-11 17:25:07 +01:00
Simon Morlat
a45d28a328
implement receiving of in-dialog chat message
2014-03-10 17:25:44 +01:00
Simon Morlat
95030951d1
add new function to play a file locally, in or out of calls.
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add new function to define a tone or wav file to be played automatically upon call errors
2014-03-04 22:58:56 +01:00
Ghislain MARY
8bc0b2e8a4
Handle RTCP XR SDP attribute parsing and creation.
2014-03-04 16:28:23 +01:00
Ghislain MARY
7e00357fd0
Fix compilation with belle-sip 1.3.1.
2014-03-04 14:32:05 +01:00
Simon Morlat
1af4a7c091
improve SDP<->SalMediaDescription conversion and offer answer algorithm
2014-02-28 16:31:05 +01:00
Ghislain MARY
ca5f624bc6
Divide huge function in smaller blocks.
2014-02-24 14:48:47 +01:00
Simon Morlat
161540a1cf
cosmetic changes
2014-02-20 15:43:32 +01:00
Simon Morlat
519430c42e
simplify creation of custom headers
2014-02-19 22:35:48 +01:00
Guillaume BIENKOWSKI
b57f8b1526
Added 302 redirection support.
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+ removed macros for sal_op_get_contact compatibility
2014-02-19 10:50:36 +01:00
Simon Morlat
9d5c1e7403
add possibility to set/get subject in SDP
2014-02-18 17:30:52 +01:00
Simon Morlat
c88c845e0c
better handling of sip chat message failures
2014-02-13 21:49:34 +01:00
Guillaume Beraudo
93f5e85967
Expose chat message reason based on LinphoneReason.
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The response code is not exposed to allow mrtp in the future.
There is no possibility to retrieve the code or the sip reason phrase.
2014-02-13 17:20:04 +01:00
Simon Morlat
931c8ade5f
Revert "Expose chat message response code and reason."
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This reverts commit f021e9aa51
2014-02-12 17:47:16 +01:00
