similar to: SIP Jitter buffer(control?)

Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use "canreinvite=yes". The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and

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]

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

[sorry for the loss of proper attributions, this is from two messages]: [Me] >This is something I've encountered in trying to make a particular > asterisk application handle properly IAX2 frames which contain either > 20ms of 40ms of speex data. For a CBR case, where the bitrate is > known, this is fairly easy to do, especially if the frames _do_ always > end on byte

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:

I get this when adding a user in ser (using serctl) root@brak sbin]# ./serctl add +18165551212 blahblah blah@blah MySql password: error: 400; check if you use aliases in SER Um error 400?? I'm lost. no docs, frustrated. venting. Matt

Any ideas? I edited the Makefile as instructed, ty. Now compiling .... sccp_channel.c 279 lines sccp_channel.c: In function `sccp_channel_send_callinfo': sccp_channel.c:48: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no

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

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

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

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

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

On 5/3/06, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> wrote: > > I must say I really like the generalized jitter buffer though :) It's a > > cleaner and more flexible implementation and can more easily be adjusted > > to contain additional information with each packet. This looks interesting to tie into asterisk's jb and plc code as well.

> Yes. Jean-Marc has made the API more similar. > > Jean-Marc: Have you looked at the API we have for the > asterisk/iaxclient jitterbuffer? Just did. > It's pretty close to what you have now -- the major difference is that > your jb still assumes it can "own" the data passed in -- it copies it, > and it destroys it at will. With the API I put together,

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

Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

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

This might be a better question for the dev list, but I don't think they want to be bothered by my silly questions. Does anyone know when we can expect to see a jitter buffer for SIP channels? I know they've been working on a generic jitter buffer since around last summer, just wondering if there's been any progress.

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 can be set to 20(msec). It is set as a Member in my class and i