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)
apt-get install sun-java6-jre
dpkg -i red5_0.7.0_all.deb
or apt-get install sun-java5-jre red5
mv red5.jar red5origin.jar
cp red5.sh red5origin.sh
cp /usr/src/red5.jar red5.jar
chmod 777 red5.jar
#put it :
exec $JAVA -Djava.security.manager -Djava.security.policy=conf/red5.policy -cp red5.jar:conf:$CLASSPATH:lib/jetty-6.1.7.jar:lib/jetty-util-6.1.7.jar:lib/jetty -xbean-6.1.7.jar org.red5.server.Standalone
mv /usr/src/sip ./webapps/sip
./red5.sh to start the server …
#if you have a problem with sip.cfg, just copy it to /usr/webapps/sip
Wow, I’ll definitely try those instructions, thanks!
On the voice quality:
red5phone converts the flash audio to G711, which is 8bits at 8kHz.
This is “regular phone” quality, minus the transcoding.
You might be able to reach phone quality by increasing the flash audio quality, and rewriting the asao2ulaw executable to handle it.
Or you can go beyond this by picking (and implementing) a higher quality codec your recieving end can handle.
BTW when the flash client runs on linux, I experience some kind of sampling rate mismatch, I sound like Alvin and the chipmunks.
No problem with windows clients.
Glad to be helpfull
And Thanks for your informations. I try to do this ASAP.
Do you know how I can change the asao rate in the asao2ulaw sources ?
- asao2ulaw reads 64 byes of asao from STDIN and converts to 256 bytes of ulaw at STDOUT
It’s a little bit hard I think …
(the sound from Flash in 8 kHZ is not really good but from the phone to flash is really better … strange …)
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.
The sound is clipped from flash windows clients (not tried yet on unix clients).
I’ve tried with many gains and silence at 10…
And yes, I’ve saw speex codec buzz (http://www.igniterealtime.org/community/message/172335) but only for flash10 and with a new version of red5 ?..
We experience no audio artefacts with windows clients.
We have ideal network conditions though (same LAN).
And use silence level 0.
Are you sure the microphone stream is not clipped to start with?
You could try the windows “media recorder” thingy, or if you know more flash than I do, you can probably do a local test recording.
The audio stream is good, I’ve testesd with many computers on differents Internet connexions with many mic…
I don’t know whith
Maybe a problem with asterisk … ??
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 did everything according to this posting under Kubuntu. Tested, able to dial the number and proceed with call but no sound !!!
When dialing there is a sound from Flex Client but later when ‘Call established’ sound dissapears and there nothing heard
on both sides. Please advice,
I’m trying to modify and compile the source code of the webphone, to meet my own requirements, without any success.
I’m using adobe flex 3.0.
am I right?
does I need anything else to compile my own?
any help will be appreciated,
Thx for a gr8 tip , i have tryed to accomplish this task but failed , can’t make any calls , do u think it’s possible to get some assistace for a fee?
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.
Please give me some idea how to run it properly.