#7 App_conference DTMF mode improvement
Hi all,
I have completed my work to improve DTMF mode. Now there is a Dummy User (reachable at 0 number) that send a video stream with a list of the user in the conference. I have made a patch that works with Asterisk 1.6 family (all version 1.6.0, 1.6.1 e 1.6.2). Unfortunately it doesn't work with Asterisk 1.4.x because there is a discrepancy in SDP negotiation (different RTP payload type).
The "video streamer" is composed by SIPp that opens SIP channel, Image Magick that draws image with the user in the conference and their ID and VLC that sends the RTP stream toward Asterisk.
I have written a INSTALL.txt to explain the installation procedure.
If you have problem with installation, please you will tell me. I think that you appreciate my work.
Best Regards
I have made an improvement of my work. Now the DTMF - DU works with Asterisk 1.6 family (all version 1.6.0, 1.6.1 e 1.6.2) and Asterisk 1.4.x. A conference room with 9 users and Dummy User requires in worst case about 6.67 Mb/s. The worst case occurs when all user use g.711a (about 90 kb/s) and display the DU (the largest bandwidth requirement is due to VLC h263p codec about 650 kb/s). I have tested the performance with Ubuntu 9.10 32 bits and CentOS 5.4 32 bits on Pentium 4 520j @ 2.8 GHz and 1 GB RAM.
PS I have also added support to iLBC codec although it doesn't work very well
Dummy User for Asterisk 1.4.x
Dummy User for Asterisk 1.6.x
Dummy User for Asterisk 1.8.x
Hi,
Now the DU mode works also with Asterisk 1.8.x. The features are the same of the previous release. Please send to me a feedback (specially to know the compatibility of hard and soft phones).