similar to: TCP port, VPN and resolving the cutting voice problem

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 hesitant to just turn on the jitter buffer in

I've been having random problems when I make long distance calls using either VoicePulse or Nufone. Sometimes the calls go through clear, and other calls (or even just part of a call) the person on the other end just hears garbled voice, or really broken up voice. Sometimes it lasts for only a few seconds, but other times it goes on for a few minutes until I give up on the call. At

I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt

Hi List; What configuration needed to let my FXS send and receive FAX? Regards Bilal

Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is intermittent break up in the audio when they are receiving. I have had a look at "iax2 show channels" and

Hi, I have a SIP client which is registered to asterisk. Asterisk is registered to a SIP trunk and also handles the media. Now since my client has some issues in its RTP Tx, which seems to have some amount of jitter (mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and max delta is 85 ms), to over come that I have enabled jitter buffer in the SIP channel by setting sip.conf

Jean-Marc Valin wrote: >>OK, I'm actually about ready to start working on this now. >> >>If people in the speex community are interested in working with me on >>this, I can probably start with the speex buffer, but I imagine >>there's going to be a lot more work needed to get this where I'd like >>it to go. >> >> > >And where

I have two IAX trunked *, there are loud crackles and pops, they are dialing out a T-1 and are sip devices, it also drops words, any help or Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060516/dc68274f/attachment.htm

> That's basically my question: the timestamps at the source and > destination are not related. Just incrementing by number of samples > doesn't really convey the real time, does it? How would a jitter > buffer know that a packet is late/early? Simple, I know what packet I just played. That gives me the "time". The jitter buffer actually makes no difference (and

Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). 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

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 in need of buffering, while channel B do need it. If A

Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards

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

> Perhaps, but then you need to assume that the jitterbuffer can just > throw away the data, and that limits how you can use it. In object- > oriented terms, you might want to pass objects to the JB, and then > call a destructor on them. In C terms, you may want to allocate > frames via malloc(), and then call free() on them later. You might > want to pass in

Jean-Marc Valin wrote: >>Heh. I guess after playing with different jitter buffers long enough, >>I've realized that there's always situations that you haven't properly >>accounted for when designing one. >> >> > >For example? :-) > > I have a bunch of examples listed on the wiki page where I had written initial specifications:

Hi List, I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that jitterbuffer is only effective on the receiving channels. My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch office. Questions: 1. To enable jitter buffer on SIP channels it seems I have to enable and force it, right? 2. If I enable and force jitter buffer, Asterisk would always have to stay

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: [asterisk-users] JITTERBUFFER function On Thu, Jan 29, 2015 at 4:56 AM,

> We just return a frame with the return value JB_DROP, which tells the > caller to drop this frame, and call jb_get again. > > When the caller is done with the jitterbuffer, it calls jb_getall() > repeatedly, until it's empty, and then it can discard all the frames. Hmm, looks a bit error-prone to me. Especially considering I still have to explain that "no, you

Jean-Marc Valin wrote: >>I believe it is adaptive, but no, I haven't used it, because it's >>coupled only to the speex codec. We're working on a generic >>application and codec-independent jitter buffer algorithm, for use in >>asterisk and iaxclient (at least). Some information is available at

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 workaround (other than jitterbuffer=off)? - Mike