search for: maxjitterbuff

...buffer=no register => username:password@parcelfarce.domain.net ;parcelfarce register => username:password@iaxtel.com ;iaxtel [parcelfarce] ;connection to parcelfarce type=friend auth=md5 secret=password context=inbound-from-parcelfarce host=parcelface.domain.net qualify=yes jitterbuffer=yes maxjitterbuffer=600 2) Set the remote jitterbuffer. I want to tell the remote Asterisk that, during this call or part of a call, that a much larger jitter buffer is OK. Basically I care more about quality of the delivered sounds, rather than latency. 3) Monitor the remote jitter buffer discards. I wan...

...39;t really matter as 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

...at is the sugested codec 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

...ain-incoming ;domain=1.2.3.4 ;allowexternalinvites=no ;autodomain=yes ;pedantic=yes ;tos=184 ;tos=lowdelay ;maxexpiry=3600 ;defaultexpiry=120 ;notifymimetype=text/plain ;checkmwi=10 ;vmexten=voicemail ;videosupport=yes ;recordhistory=yes disallow=all allow=g729 allow=gsm allow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = defaul...

...(number)}) exten => s,n,Background(Outsource) exten => s,n,WaitExten(10) exten => s,n,Goto(inside,133,1) exten => 9,1,Background(OEM_Menu) exten => 9,n,WaitExten(10) exten => 9,n,Goto(0,1) exten => 0,1,Goto(inside,133,1) IAX.conf [general] jitterbuffer=yes forcejitterbuffer=no maxjitterbuffer=500 autokill=yes ; --------------------------------------------------------- ; IAX INCOMING USER ; ; This is the user for incoming calls from: ; connect02.voicepulse.com ; --------------------------------------------------------- [voicepulse] ; <-- Name m...

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: >>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>>> trunk between them. > Carlos, > > Had caller-id ever worked between these two systems? > > Doug > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

...2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel -------...

...conversion. sip.conf... --- [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)...

...Best: 99.999 -- Worst: 99.993 -- Average: 99.996930, Difference: 99.997334 <br> <br> ; iax.conf server_master <br> ; <br> [general] <br> bindport=XXXX <br> bindaddr=222.222.222.222 <br> delayreject=yes <br> language=fr <br> allow=alaw <br> maxjitterbuffer=800 <br> trunktimestamps=yes <br> tos=ef <br> <br> [jourdain] <br> type=friend <br> host=111.111.111.111 <br> port=XXXX <br> context=iax-jourdain <br> trunk=yes <br> disallow=all <br> allow=alaw <br> jitterbuffer=no <br&gt...

...w----> Asterisk #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...

...onf files on each end start like 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 extr...

...an_sip.c:5860 handle_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 =...

...th). We've no E1/T1 links, everything is IP based. My boss 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...

...|s|1) iax.conf file -------------------------------------------------------------------------------------- ; 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 [o...

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...r is using ztdummy. 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 binda...

I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls.

...obal default as to whether you want ; the jitter buffer at all. ; ; dropcount: the jitter buffer is sized such that no more than "dropcount" ; frames would have been "too late" over the last 2 seconds. ; Set to a small 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...

...osted below: type=friend trunk=yes allowcallerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=100 resyncthreshold=1500 tos=ef cos=5 Doug

...%0 33.82 42.88 1904.79 36.62 131919 %0 0.00 47.04 1924.64 36.36 Using these numbers, and knowing that the max rtt will not happen very often how do the jitter settings below look? Does anyone have any recommendations to improve my call quality using the jitterbuffer? jitterbuffer=300 dropcount=1 maxjitterbuffer=500 maxexccessbuffer=20 -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643