similar to: Jitter buffer for SIP channels (OT?)

Hi, I really want to start using 1.2.5, but I also really need to have the jitter buffer. Can anyone offer a suggestion of how to go? I've looked at the SIP_JB patch and there seems to be no indication of a patch for the 1.2.5 release.

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

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.com ;iaxtel [parcelfarce]

Danny Chan wrote: >Hi Jean and Steve, > >Can you tell me whether the jitter filter / buffer is adaptive type, I >saw the description of speex_jitter.h say it is "adaptive", anyone of >the group has implemented it and confirm it. > > 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

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:

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

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

We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want

> 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, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter

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'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

Dear speex developers and users, I'm considering adopting the speex jitter buffer for use with a different codec in a voice conferencing system and would be very grateful if those more acquainted with it could help me with some questions. The speex_jitter_buffer.c wrapper seems to maintain (buffer?) one packet-frame ("current_packet") in addition to the packets already

Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My

Hi! I just wondered if anyone would mine posting their successful jitter buffer settings here for me if they get a moment ?? I've spent a few hours trying to set the jitter buffer up reasonably logically and can definitely tell it makes a difference and can introduce latency and echo if setup incorrectly but I can't see a good post anywhere describing properly what the three settings

Thanks for your reply Jean-Marc! this was what I had before. But I decided to restructure it since the thread that plays the sound is a callback from the sound hardware, more or less an interrupt 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

Jean-Marc Valin wrote: >>In particular, (I'm not really sure, because I don't thorougly >>understand it yet) I don't think your jitterbuffer handles: >> >>DTX: discontinuous transmission. >> >> > >That is dealt with by the codec, at least for Speex. When it stops >receiving packets, it already knows whether it's in DTX/CNG mode.

Hello. I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post ( http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) I added following scheme

According to the wiki, "IAX2 jitter buffer (when turned on) doesn't currently work well with trunking (trunk=yes in iax.conf) Yourcall...sstartto....soundlike....this :-)" Is this still the case? If that's so, how do you do to use trunking in conjunction with a PRI card? Do you use another, separate asterisk box for the jitter buffer? Cheers, Jean-Michel.

Hi Folks, I know the PAP2T-NA has a jitterbuffer.... however, it seems to be adaptive, which is fine for most situations... however, is there some way I can either: A) Specify how long it waits before it starts to shrink? B) Specifiy a fixed sized jitterbuffer? -------------- next part -------------- An HTML attachment was scrubbed... URL: