Asterisk Connector Bridge

ckleea

Anyone is interested in this? I am looking for possibility to build a better TTS for my asterisk server.

Asterisk Connector Bridge 0.3.0 Released.

unimrcp Media Resource Control Protocol (MRCP) allows to control media processing resources on the network using a distributed, client/server architecture. The main media processing resources specified by the MRCP standard are

. Speech Synthesizer (TTS)
. Speech Recognizer (ASR)
. Speech Recorder (SR)
. Speaker Verifier (SV)

MRCP is not a standalone protocol and it relies on various VoIP protocols such as

. SIP (MRCPv2), RTSP (MRCPv1) session management
. SDP offer/answer model
. RTP/RTCP media streaming

UniMRCP is an open source cross-platform MRCP project, which provides everything required for the implementation and deployment of both an MRCP client and an MRCP server. UniMRCP encapsulates SIP/MRCPv2, RTSP, SDP and RTP/RTCP stacks and provides integrators with MRCP version independent user level API.

Everybody is welcome to join the community, use and make the project better by participating in discussions, raising issues, providing patches.

The connector bridge is prepackaged with the latest Asterisk-1.6.2.9 and UniMRCP-r1744 (> 1.0.0). However, previous versions of Asterisk and UniMRCP are supported as well. This release contains several enhancements in both res_speech_unimrcp and app_unimrcp modules.

Changes in res_speech_unimrcp include

. Made an enhancement to SpeechLoadGrammar to be able to specify an input grammar as a URI too. (Raymond)
. Fixed compilation of res-speech-unimrcp module for Asterisk 1.4.
. Fixed processing of Set-Input-Timers header field.
. Set an interpreted result based on the element instead of the one.

Changes in app_unimrcp include

. Changes required for version 1.2 of Asterisk (Issue-64, Igor, Derik)
. Added missing '{' to compile with the released UniMRCP version too. (Issue-65, Igor)
. Bug fix to check if codec descriptor could be obtained (Derik)
. Added support for ABNF grammar (Issue-76, Assanta, Derik)
. Bug fix to speech_channel_destroy (Issue-72, Assanta, Derik)
. Addition of request-timeout configuration parameter (Derik)
. Bug fix to address issue 80  - checking for speech channel state while waiting for audio frames in MRCPRecog (Assanta, Derik)
. Removed incorrect check for resf which was fixed at -1 anyway (Assanta, Derik)
. Added SYNTHSTATUS and RECOGSTATUS variables so that problems can be detected in the dialplan (Assanta, Derik)

The released package can be downloaded from
http://unimrcp.googlecode.com/files/uni-ast-package-0.3.0.tar.gz
For the installation, configuration and usage please refer to the wiki page
http://code.google.com/p/unimrcp/wiki/asteriskUniMRCP

bubblestar

Thanks for the information.  It is great.

It seems they have no demo provided in the site.  Don't know how it sounds like.