OpalVOIP 3.6.3

OpalVOIP is a C++ multi-platform, multi-protocol library for Fax, Video and Voice over IP and other networks.
Opal runs on Mac OS X, Linux, Windows, Solaris, xBSD and Windows Mobile.
NOTE: OpalVOIP is licensed and distributed under the terms of the Mozilla Public License 1.0 (MPL).

Main features:
- Low latency RTP stack designed specifically for real-time multimedia
- Full featured H.323, SIP and IAX2 protocol stacks
- Audio codecs including G.711, GSM06.10, Speex and iLBC.
- Video codecs including H.261 and H.263
- Run-time loadable codec interface for proprietary or codecs such as G.729, H.263, H.264 and MPEG4
- Completely Open Source using the commercially friendly MPL 1.1 license

System requirements:
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- PTlib

Enhancements
- Fixed race condition in silence insertion algorithm where silence buffer is zero length before the first packet is received. Some downstream channels cant handle this. Now initialise silence buffer to be at a minimum 10 milliseconds.
- Fixed compile without video
- Backport of 22121 from trunk Reintroduce implicit check for NULL buffer accidentally removed in revision 22089
- Fixed empty Alert-Info field being sent in SIP INVITE.
- Fixed display name member of OpalConnection containing ONLY the display name for SIP.
- Fixed call routing issue in simpleopal
- IF we use string options to override the username (Calling-Party-Number) then we override it, no ifs buts or maybes!
- Fixed incorrect detection of remote SIP client putting local side on hold, thanks hongxing.
- Fixed incorrect trace log messages.
- Fixed fax NSE tone detect of CNG/CED being passed up to connection and application, for some reason commented out! Fixed OpenPhone detection that fax mode is already in use when get a CNG/CED indication.
- Fixed switching media format for existing channel, e.g. T.38 mode, was broken by someone.
- Slight change to routing algorithm so if the B-Party is explicitly determined, even when there is no matching "source" route in the table.