Merged revision(s) 31686-31696 from opal/branches/v3_14:
---------------------
r31696
Fixed setting of audio increment counts and tx buffer sizes for "special" codecs, e.g. G.722.
---------------------
r31695
Fixed G.722 plugin codec, incorrect samples/second which is really the RTP frame time in this context.
---------------------
r31694
Fixed use of correct flag for a streamed audio plug in codec.
---------------------
r31693
Another fix for too early call to OnStartMediaPatch when using external RTP.
---------------------
r31692
Tidied trace log of "keep-alive" packets from some gatekeepers that send four zero bytes stream of CRL/CRLF which is what the standards say.
---------------------
r31691
Fixed (yet again) issue when a registrar returns a Contact fields we did not register, during the unregister process.
---------------------
r31690
Fixed output of media stream details in OpenPhone status window.
---------------------
r31689
Fix for too early call to OnStartMediaPatch when using external RTP. The OpalDummySession needs to have a remote media address before it can be considered "open".
---------------------
r31688
Factored output of OpalMediaStream details to ostream, done in several places, now one.
---------------------
r31687
Fixed (again) issue when a registrar returns a Contact fields we did not register, during the unregister process. Unusual, but perfectly legal.


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31697 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31697-31704 from opal/branches/v3_14:
---------------------
r31703
Yet another fix for too early call to OnStartMediaPatch when using external RTP.
---------------------
r31702
Added VisualStudio 2012 debugging "visualizer" file for OPAL variables.
---------------------
r31700
Fixed using local media options as base for remote H.323 capabilities, this fixes issue where our send video resolution is not adjusted to value in AdjustMediaOptions() as they should.
---------------------
r31699
Fixed SDP imageattr remote send resolution affecting our send resolution.
---------------------
r31698
Fixed GNU compile 


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31705 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31705-31707 from opal/branches/v3_14:
---------------------
r31707
Fixed crash in FFMPEG (e.g. H.264) decoder when get an empty RTP packet before first decoded frame.
---------------------
r31706
Tweak to ARQ so if an explicit IP:port is used to make a call, only the destCallSignalAddress is filled in and the destinationInfo is not. Confuses some gatekeepers,


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31708 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31708-31710 from opal/branches/v3_14:
Fixed issues with recording to AVI file (Windows only) supporting DivX codec.


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31711 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31711-31726 from opal/branches/v3_14:
---------------------
r31724
Updated G.722.1 plug in codec to libg722_1 code set, including support for G.722.1 Annex C operation.
---------------------
r31723
Fixed codec test application in audio mode.
---------------------
r31720
Fixed setting of H.323 gatekeeper RAS connections local interface/port. Added H.323 gatekeeper interface/port to OpenPhone options dialog.
---------------------
r31719
Fixed mistaking an E.164 number as an IP address.
---------------------
r31718
Fixed accidental check in of licensed build for x264. Default should be GPL version.
---------------------
r31715
Fixed linker warning.
---------------------
r31713
Make sure RTP port range is always even.


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31727 023b2edf-31b2-4de3-b41e-bca80c47788f
Addition of DTLS version of SRTP, big thanks to the initial code from Sysolyatin Pavel. All current breakages are mine.

git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31753 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31727-31753 from opal/branches/v3_14:
---------------------
r31751
Fixed error path that ends up access NULL transport in SIP calls.
---------------------
r31749
Applied patch #273 Add IsCompatibleTransport methods to WS & WSS transports, thanks  Valeriy V. Argunov
---------------------
r31743
Workaround for GnuGK, have to indicate H.460.18 support on every RRQ, spec says don't have to on keep alives.
---------------------
r31739
Removed assert if could not create an AVI codec in recording.
---------------------
r31738
Attempt to fix issue where if both C# and Java generated file does not exist, you have to build twice for it to generate them both. Seems to be a weird bug in VisualStudio, how these two completely different build steps become entangled is ...
---------------------
r31737
Use internal image inversion for AVI recording, rather than PVideoConverter.
---------------------
r31735,r31734
Fixed SIP offer of G.722.1 Annex C variants.
---------------------
r31731
Fixed use of pkg-config with OPALDIR set. 
---------------------
r31730
Moved libg722_1 configure from Makefile to configure.ac where it belongs, and more importantly, works. 
---------------------
r31729
Added executable attribute to scripts in G.722.1 lib.
---------------------
r31728
Fixed Linux build of new G.722.1 plug in codec. 


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31755 023b2edf-31b2-4de3-b41e-bca80c47788f
Fix compilation without SRTP

git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31757 023b2edf-31b2-4de3-b41e-bca80c47788f
Update configure script


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31759 023b2edf-31b2-4de3-b41e-bca80c47788f
Fixed compile with RTP FEC enabled


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31760 023b2edf-31b2-4de3-b41e-bca80c47788f
Fixed DTLS code compatibility with older GNU compiler


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31761 023b2edf-31b2-4de3-b41e-bca80c47788f
Fixed 64 bit compile and compiler warning.

git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31765 023b2edf-31b2-4de3-b41e-bca80c47788f
Start of registrar implementation for OPAL server, not this is not a full proxy, just a registrar.

git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31781 023b2edf-31b2-4de3-b41e-bca80c47788f
Applied patch #275 Some fixes for DTLS, thanks Sysolyatin Pavel

git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31783 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31754-31783 from opal/branches/v3_14:
---------------------
r31776
Make sure Target Bit Rate set in audio Media Format from plug in.
---------------------
r31774
Moved multiple gatekeeper monitor threads to single one when have alias limit on registrations. Make sure endpoint is re-registered to gatekeeper when aliases or alias patterns are changed.
---------------------
r31773
Make sure gatekeeper identifier updated when no GRQ used. Use one in RCF.
---------------------
r31772
Fixed issue with closing down call when H.460.18 started separate H.45 channel.
---------------------
r31771
Fixed missing H.460.18 keep-alive on H.323 signalling and control channels.
---------------------
r31770
#274 Quick fix for the bug #508 Assertion on receiving SIP response for the SIP BYE transaction, thanks Valeriy V. Argunov
---------------------
r31769
Fixed H.460.18 handling of H.245 secondary TCP stream.
---------------------
r31768
Added separate configurable time for NAT pin-hole keep-alive and change H.323 RAS transport to use it to send empty TPKT (as per H.460.18) if registration time to live longer.
---------------------
r31767
Fixed setting of H.323 RAS transport interface in other than just GCF. Needed when GRQ is suppressed.
---------------------
r31766
Added "Suppress GRQ" to H.323 gatekeeper options in OpenPhone
---------------------
r31763
Fixed compiler warnings 
---------------------
r31762
Fixed compile of internal libSRTP on *nix 


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31784 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31784-31789 from opal/branches/v3_14:
---------------------
r31789
Added RAS channel network interface option to console user interface. 
---------------------
r31788
Added suppress GRQ option to console user interface.
---------------------
r31785
Fixed gatekeeper destination being NAI derived from URL.


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31790 023b2edf-31b2-4de3-b41e-bca80c47788f
Applied patch #276 Another fixes for DTLS, thanks  Sysolyatin Pavel

git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31813 023b2edf-31b2-4de3-b41e-bca80c47788f
Merged revision(s) 31790-31815 from opal/branches/v3_14:
---------------------
r31814
Applied patch #277 Added x64 configurations for vp8_webm plugin. Thanks Sysolyatin Pavel
---------------------
r31811
Fixed build when OPAL_LID disabled.
---------------------
r31808
Added local ringback tone generation for PCSS connection.
---------------------
r31807
Make sure OnStartMediaPatch called on OpalConnection for both sources and sinks.
---------------------
r31806
Added WAV file play to local sound device on OpalConsolePCSSEndPoint (conopal)


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31817 023b2edf-31b2-4de3-b41e-bca80c47788f
Update symbols

git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31822 023b2edf-31b2-4de3-b41e-bca80c47788f
Fixed detecting internal Opus codec build on *nux


git-svn-id: svn://svn.code.sf.net/p/opalvoip/code/opal/trunk@31826 023b2edf-31b2-4de3-b41e-bca80c47788f