search for: maxexcessbuff

...odec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David

...onf... --- [sip-router-1.gradwell.net] context=sip-inbound type=peer host=sip-router-1.gradwell.net [sip-router-2.gradwell.net] context=sip-inbound type=peer host=sip-router-2.gradwell.net --- iax.conf... [general] bandwidth=high disallow=lpc10 jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=lowdelay --- when a call comes in, we dial something like this, in our dial plan: -- Executing Goto("SIP/213.166.5.134-118f5310", "sip-users|7770002|1") in new stack -- Goto (sip-users,7770002,1) -- Executing Dia...

...#1 <------IAX (usually GSM)---------> Asterisk #2 <-------- IAX (usually GSM) --------> Asterisk #3 <------ulaw-----> softphone. Now originally I had all jitter buffers turned off. Today I tried turning the jitter buffers for Asterisks #1 and #3 on with maxjitterbuffer and maxexcessbuffer set to 1000. That seemed to help a little. Then I tried those settings for Asterisk #2 and it seemed the same or worse. I have tried several different audio codec choices as well. The puzzling thing is, the video works pretty well but the audio has the trouble. It's largely bandwidth-r...

...l number. "3" represents 1.5% of frames dropped ; ; maxjitterbuffer: a maximum size for the jitter buffer. ; Setting a reasonable maximum here will prevent the call delay ; from rising to silly values in extreme situations; you'll hear ; SOMETHING, even though it will be jittery. ; ; maxexcessbuffer: If conditions improve after a period of high jitter, ; the jitter buffer can end up bigger than necessary. If it ends up ; more than "maxexcessbuffer" bigger than needed, Asterisk will start ; gradually decreasing the amount of jitter buffering. ; ; minexcessbuffer: Sets a desired mim...

...e this: > [general] > trunk=no > notransfer=yes > iaxcompat=no > > bandwidth=low > > disallow=all > allow=ilbc > > jitterbuffer=yes > dropcount=3 > maxjitterbuffer=500 > maxexcessbuffer=150 > minexcessbuffer=40 > jittershrinkrate=1 Of course, perhaps the jitter buffer isn't to blame, but given that one side of the call sounds perfect, I can't think of anything else obvious that would cause this. Is there any way to extract from asterisk some idea of...

...response: Peer '6666' is now REACHABLE! Iax.conf: [general] ;port=5036 ;I commented this out because I discovered it is hard coded anyway. bindaddr=10.20.30.10 amaflags=default accountcode=default bandwidth=low disallow=all allow=gsm jitterbuffer=no ;dropcount=3 ;maxjitterbuffer=500 ;maxexcessbuffer=100 ;trunkfreq=20 tos=lowdelay authdebug=no register => username:password@xxx.xxx.xxx.xxx ;these are real values that I changed for security reasons [arlington] type = friend context = local callerid = auth = plaintext secret = password ;inkeys = ;outkey = host = dynamic ;defau...

...ss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file has: disallow=all allow=ulaw allow=alaw bandwidth=high jitterbuffer=yes dropcount=2 maxjitterbuffer=1000 maxjitterinterps=10 resyncthreshold=1000 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=lowdelay autokill=yes He complains of broken audio, muffled audio, and says compared to Skype its very poor, particularly during conference calls (zaptel meetme). Most of these would be SIP based within our server though, rather than IAX/PSTN based (X...

...e -------------------------------------------------------------------------------------- ; iax.conf [general] ${INCOMING-USR}=SECRET-USERNAME ${INCOMING-PWD}=SECRET-PWD ${LIVEVOIP-SVR}=217.160.244.186 bandwidth=high disallow=lpc10 jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 register => ${INCOMING-USR}:${INCOMING-PWD}@${INCOMING-SVR} tos=lowdelay [incoming] ; this is the incoming IAX provider type=user secret=ITS-SECRET deny=0.0.0.0/0.0.0.0 permit=217.160.244.186/255.255.255.0 context=arbitrary-in [outgoing] ;this is t...

...there's no buffer for SIP) as recorded by 'sip show peers' as about 70ms (but in reality ping I think is about half this). My current setup is : SIP and IAX clients --> my * --> providers via IAX (one 5ms away, one 80ms away) jitterbuffer=yes dropcount=5 maxjitterbuffer=100 maxexcessbuffer=45 If anyone can post any amplification on these settings apart from what's in iax.conf or their experiences or maybe some adjustments I should try that'd be really really helpful. Thanks!! Chris

...IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes bandwidth=low disallow=ilbc disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=ulaw ;allow=gsm ; Always allow GSM, it's cool :) jitterbuffer=no trunkfreq=20 ;dropcount=3 ;maxjitterbuffer=500 ;maxexcessbuffer=100 ; tos=lowdelay register => lachnet@woodlane.lach.net register => lachnet@10.1.1.158 ; [woodlane] allow=ulaw ;allow=gsm type=friend jitterbuffer=no username=woodlane context=dialout host=dynamic trunk=yes trunkfreq=20 IAX.conf on machine2: [general] port=5036 bindaddr = XXX.XXX.XXX.XXX...

...the output) and am then able to talk both ways for 20 to 30 seconds and then the IAX phone appears off line. If I wait on the PSTN line for another 30-45 seconds, I get the Hungup lines in the trace. I've tried jitterbuffer=no and then jitterbuffer=yes with dropcount=3 maxjitterbuffer=500 maxexcessbuffer=100 (defaults in the iax.conf) and neither setting worked. What is wrong? TIA for any help!!!!!! Ben Here is the console output: -- Accepting AUTHENTICATED call from 192.168.1.103, requested format = 2, actual format = 2 -- Executing Dial("IAX2[2002@2002]/1", "IAX2...

...cho when going out an FXO port through a zaptel channel. It wasn't obvious to me if this same process could be done on the IAX channels and if so how? For reference my iax.conf looks likes: [general] port=5036 bindaddr=0.0.0.0 bandwidth=low echocancel=yes jitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=100 dropcount=5 register => djohnson:12dookie@iaxtel.com tos=lowdelay #include /etc/asterisk/users/iax/iax_users [iaxfwd] type=user context=IAX_FWD deny=0.0.0.0/0.0.0.0 permit=65.39.205.0/255.255.255.0 Thanks for the help! dj

...secion. Should I change the bindaddr? I thought 0.0.0.0 just listened to everything, so that's why it did that. [general] port=4569 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 allow=g726 iaxcompat=yes bandwidth=high jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxexcessbuffer=300 tos=0x18 register => XXX:XXX@voip.teliax.com IAX2 debug shows so much info I don't know where to start. I don't see any errors or warnings in the console though. TIA, -Ron

...iax.conf the audio delay problem is solved, but the call quality degrades. My prior settings were: [general] bindport = 4569 bindaddr = 0.0.0.0 delayreject=yes disallow=all allow=ulaw allow=alaw mailboxdetail=yes register => XXXXX@voip.teliax.com dropcount=3 jitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=300 canreinvite=no Any ideas on tweaking iax.conf to optimize call quality, but avoid the audio delay with forwarded calls? Regards, Mike Michael Hillerbrand

...tune codecs here using "allow" and "disallow" clauses ; with specific codecs. Use "all" to represent all formats. ; disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=gsm ; Always allow GSM, it's cool :) jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=10 jittershrinkrate=1 register => XXXXXXXX:xxxxxxx@iaxtel.com ; Finally, you can set values for your TOS bits to help improve ; performance. Valid values are: ; lowdelay -- Minimize delay ; throughput -- Maximize throughput ; reliability -- Maximize reliability...

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

...Config files (colo* and office* have identical configs except where noted): iax.conf: [general] disallow=all allow=ulaw allow=gsm tos=0x18 pingtime=1 lagrqtime=1 ; tried iaxcompat=no and yes, no apparent difference iaxcompat=yes ;delayreject=yes jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=25 jittershrinkrate=1 (colo* has an entry for office-ast, office* has an entry for colo-ast with the host entries set respectively) [office-ast] type=peer host=192.168.2.1 qualify=500 trunk=yes [phone] type=user context=incoming secret=***** host=192.168.2.1 qualify=500 dis...

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Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm