search for: jitterbuff

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not...

WTF is a jitterbuffer? Sent from my Verizon Wireless 4G LTE smartphone -------- Original message -------- From: Matthew Jordan <mjordan at digium.com> Date: 01/29/2015 10:41 AM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re:...

In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share

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 i...

...miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only jitterbuffer=no (and * restart), recorded VM messages are very clean. With jitterbuffer=yes and trunk=no, messages are very clean. Both boxes config'ed as: trunk=yes allow=gsm jitterbuffer=yes Recorded VM messages are very clean....

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the jitter > bu...

I know, that jitterbuffer should be set at receiving side and on outgoing call leg, ie. if sipphone calls to asterisk and outgoing to zap chanel, I should set jitterbuffer on zap channel (to dejjitter audio stream from sipphone) but what about pure voip situation (i.e. iax-iax, sip-iax, skinny-iax etc.)? I have followi...

Hello, After checking out CVS HEAD from yesterday (for those new PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom IP600's. After seing it resolved as of this morning (thanks Mark), I decided to try again... I can answer incoming calls. No problem there. Putting calls on hold, however, results in my Polycom IP600 indicating the call on hold, but...

On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson <torbjorn.abrahamsson at gmail.com> wrote: > Hello! > > > > I am going to use the JITTERBUFFER function in a SIP (and local channels) > only setup, but have some questions of how to use it: > > > > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > the JITTERBUFFER function? You only need to use the JITTERBUFFER function. The jbenable option...

Hi, I am using the JitterBuffer. Since there is not so much documentation I think I dont use it in a correct way. All the packets are recieved (I control the sequence numbers) but the JitterBuffer often tells me he has no packet. I am using it in the following way: I am not sure if I use the ticks correctly but I think it c...

Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having the same symptoms (i.e: no voice...

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaro...

...-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of yusuf Sent: Monday, February 20, 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Adam Robins wrote: > Hi Adam > After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls "breaking up" from both the customer and agent sides. What...

I've been having problems with incoming IAX2 calls - some work, but a large fraction are answered with "dead air" or disconnects from my IAX provider. Disabling the jitterbuffer seems to eliminate the problem (so far)! Has anyone else seen this? I'm using 1.2.6, but I'm not sure what my provider is using. A snippet of the a failed incoming call IAX2 debug is attached below (with jitterbuffer on). Note the HANGUP and INVAL codes. - Mike Rx-Frame Retry[ No...

...rupt handler. For me it seems more reasonable to waste some memory for to save the decompressed Packet. While I write this I begin to think that it is possible I decompress Packets that are never used because they are too late. What is the reason why one should put the compressed Packet in the JitterBuffer? David > (Sorry about the delay -- currently attending ICASSP) > Hi, > > Haven't looked at all the details, but what's clearly wrong is that you > need to put the *compressed* packets in the jitter buffer and decode > them only when you _get() them. > > Je...

What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a "PSTN" server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are having latency and jitter issues. I am h...

Hi, I have a question about jitterbuffer and PLC. I use Asterisk 1.6.2.0 and 1.6.0.20 or older. I use uLaw. My system map: ============================================================================= [ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ] =======================================================...

...occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We have enough bandwidth, the latency to the servers is 100-150ms and the packet loss is around 1%. We tried using the G711 and G729 codec and also have the jitterbuffer enabled. How can I solve this problem of voice quality? Can a better implementation of jitterbuffer with packet loss concealment help? If so how do I get the newer implementation. I would really like to help out in the development of the new jitterbuffer if it has not yet been implemented....

Are these things possible? 1) Set the local Asterisk jitterbuffer size, but only for a particular connection. I'd like to force Asterisk to use a particularly large buffer in certain cases. Should I expect this to work? [general] jitterbuffer=no register => username:password@parcelfarce.domain.net ;parcelfarce register => username:password@iaxtel...