search for: minexcessbuffer

...] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw notransfer=yes trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 I have tried with jitterbuffer=no, and then rather then one-way-audio I get high packet loss until the connection settles back down. Any ideas on other things I can try?

...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

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255 1207 OK (33 ms) Why is it running on port 1207?

...outer-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 Dial("SIP/213.166.5...

...> 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 why it thinks the calls sound bad...

...any 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-lite/SJphone). Obv...

...--------------------------------------------------------------------- ; 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 the outgoing IAX provi...

...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 mimimum amount of headroom in ; the jitter buffer. If Asterisk has less headroom than this, then ; it will start gradually increasing the amount of jitter buffering. ; ; jittershrinkrate: when the jitter buffer is being gradually shrunk ; (or enlarged), how many millisecs shall w...

...ng "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 ; mincost -- Minim...

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

...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 disallow=ulaw trunk=yes...