Yann Diorcet
a87c70e44a
Improve uPnP API and uPnP mechanism
2013-01-09 11:56:55 +01:00
Yann Diorcet
33bab1941e
Improve uPnP api
2013-01-09 10:28:18 +01:00
Yann Diorcet
ce87dab637
Add message on port mapping update/clean
2013-01-08 17:30:32 +01:00
Yann Diorcet
cc592dec5d
Add better uPnP description
2013-01-08 17:28:01 +01:00
Yann Diorcet
215b566b2c
Add debug log on port mapping add/remove send
2013-01-08 17:15:46 +01:00
Yann Diorcet
92c9faec6e
Improve uPnP
2013-01-08 17:06:27 +01:00
Yann Diorcet
4e90f134d5
Add another early port binding release
2013-01-08 14:33:48 +01:00
Yann Diorcet
492f3c9b91
Update upnp igd, and early destroy upnp session on call fail
2013-01-08 13:56:07 +01:00
Yann Diorcet
4b257d3de4
Don't remove hook (done before)
...
Remove unused variable
2013-01-07 16:19:20 +01:00
Yann Diorcet
92a7d6695c
Set external port equal to local port the first time
2013-01-07 16:08:09 +01:00
Yann Diorcet
9c3097ab3d
Fix invalid port binding comparaison
2013-01-07 09:50:53 +01:00
Yann Diorcet
f3805137e6
Working call with uPnP
2013-01-04 16:19:13 +01:00
Yann Diorcet
9567e2bf62
Working sip upnp
2013-01-03 15:47:38 +01:00
Yann Diorcet
806203ca0a
uPnP in progress
2012-12-21 16:21:41 +01:00
Yann Diorcet
8026b597a7
Starting uPNP integration
2012-12-21 10:11:06 +01:00
Yann Diorcet
9898c4beec
Fix static build split tunnel stubs
2012-12-20 17:34:36 +01:00
Yann Diorcet
deccbd6963
Start including upnp
2012-12-19 15:41:23 +01:00
Sylvain Berfini
d03b0d01be
Remove typo error
2012-12-14 16:36:24 +01:00
Sylvain Berfini
320111b1a4
Fix enable low bandwidth jni interface
2012-12-14 16:29:03 +01:00
Yann Diorcet
148a0a91b5
Add XML2LPC lib/tool
2012-12-13 16:08:43 +01:00
Simon Morlat
303b8e5d6c
better proxy & route management.
2012-12-13 14:50:09 +01:00
Sylvain Berfini
58c708a10b
Add JNI glue to get timestamp for logs
2012-12-11 17:00:20 +01:00
Sylvain Berfini
1e75dc4022
Add JNI glue to enable/disable low bandwidth param in call
2012-12-11 10:59:40 +01:00
Yann Diorcet
9593053c8d
Merge branch 'master' of git.linphone.org:linphone
2012-12-11 10:40:05 +01:00
Ghislain MARY
7d07ca75e7
Apply the user agent as soon as it is changed.
...
The user agent is no longer global and now depends on the linphone core
(kind of, there is still a global variable behind to hold its value).
2012-12-11 10:38:57 +01:00
Yann Diorcet
7f5682a95d
Revert "Fix shell substitutions"
...
This reverts commit 7095521237 .
2012-12-11 10:03:16 +01:00
Yann Diorcet
7095521237
Fix shell substitutions
2012-12-11 09:29:53 +01:00
Yann Diorcet
3e7a110e01
Fix autotool warnings
2012-12-10 17:03:07 +01:00
Ghislain MARY
4821958ecd
Improve audio hack for Galaxy S.
2012-12-10 16:03:44 +01:00
Ghislain MARY
0849cfcffa
Fix typo in getRemoteUserAgent() method.
2012-12-10 09:41:49 +01:00
Guillaume Beraudo
ba478b89e8
Fix crash on INVITE without SDP.
2012-12-04 16:23:07 +01:00
Ghislain MARY
d7cf616e07
Add getRemoteUserAgent() to the JNI.
2012-12-04 14:04:33 +01:00
Simon Morlat
43c255f7fb
fix low bandwidth mode
2012-12-03 14:48:05 +01:00
Simon Morlat
8648c08a0d
update ortp and ms2, and set rfc2833 the default
2012-11-29 17:21:27 +01:00
Sylvain Berfini
07c4120600
Fix typo mistake
2012-11-29 16:46:33 +01:00
Sylvain Berfini
019c6f1c55
Added SIPINFO/Rfc2833 JNI glue
2012-11-29 16:40:14 +01:00
Simon Morlat
c6bd038a6d
implement manual low banwdwidth mode.
...
It is also possible to check whether peer is under low bandwidth by looking into the linphone_call_get_remote_params()
2012-11-27 14:45:02 +01:00
Yann Diorcet
17c6593abc
Add getter for zrtp_secrets_file, static_picture and root_ca
2012-11-26 10:51:50 +01:00
Jehan Monnier
501c5ce4db
add function to extract ccc from e164 number
2012-11-23 10:38:15 +01:00
Simon Morlat
96dc7aac83
write documentation of audio conferencing module.
2012-11-22 23:11:13 +01:00
Simon Morlat
d0745a39f2
better srtp management
...
- have the choice to keep same keys accross reINVITEs
- don't restart the stream for minor changes like removal of a recv-only codec.
2012-11-22 22:02:32 +01:00
Ghislain MARY
06d8cec790
Add checks to prevent crashes.
...
The getReceiverInterarrivalJitter() function may be called before the
used audio codec is known in case of early media, so test that the
payload type is valid before using it.
2012-11-22 11:36:29 +01:00
Yann Diorcet
362d77908a
Call zoom video: check output nullity
2012-11-21 12:16:17 +01:00
Jehan Monnier
2c5421f22e
add more dial plan support
2012-11-19 15:59:34 +01:00
Jehan Monnier
fb9fa2aa00
fix for lc->last_recv_msg_ids size limit
2012-11-19 15:59:34 +01:00
Sylvain Berfini
eeb4b52d1e
JNI glue for linphone_core_set_primary_contact
2012-11-16 14:20:59 +01:00
Simon Morlat
7b95eead97
sip dscp wasn't taken into account immediately.
2012-11-12 21:26:29 +01:00
Simon Morlat
bcfc9a02e0
update ms2
2012-11-12 12:22:20 +01:00
Ghislain MARY
72f307c556
Fix compilation if git is not installed.
2012-11-09 10:49:15 +01:00
Sylvain Berfini
2123621e56
Fix microphone gain db
2012-11-09 09:40:32 +01:00