First, thanks for red5phone, it’s amazing !!! nice job !!!
I’ve got an ubuntu server with a standalone SIP red5phone. It’s working !!
The sound from flash isn’t good, how can I improve it ? (I’ve tried to put 44 in flex and 44000 in sip.cfg without any success, maybe I’ve to change audio_frame_size too ?). I think I need to edit java sources but where
Thanks a lot
To make it works under ubuntu :
(This is my installation on a clean Ubuntu 8.04 server)
Sorry, no, I was just outlying the theoretical possibilities based on my understanding of the architecture and the protocols involved, but I never dabbled in codec implementation myself.
I also wanted to point out that speech quality did not seem bad to me at all on windows clients, and the distorsion on linux clients sounds more like a sampling rate issue than a quality issue.
Are you sure your microphone gain levels are set right? Have you tried with echo cancellaton on and off? What kind of quality degradation do you experience?
And finally I know this is definitely not ASAP, but the speex codec is coming in flash plugin 10, check out the buzz on the red5 mailing list.
You can rule out astersik and the other endpoint by taking a network capture and listening in on the G711 transmission in WireShark (formerly ethereal).
It’s simplest to take the capture on the machine running red5, either by wireshark directly, or by tcpdump (wireshark can open tcpdump dumps.)
Then in the Statistics menu there’s a VoIP entry, allowing you to “play” the selected call stream by stream.
I download SIP Phone but now I don’t know how to configured it in flex and run it. I import Flex file in flash builder and when I run the program it show me Login page and when I fill up detail it’s show me connection failde.