similar to: "jittershrinkrate" equivalent in current (new) iax jb implementation

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 following setup (homeworkers using sip phone connected to home asterisk via SIP and

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the remote facilities and try to call the G3R's voice mail system(Audix), DTMF is not recognized unless

Ah, I'm sorry, I have read the manual and believe I have a reasonably good grasp on how to use the Speex encoder and decoder altogether. In fact I've been using it with great success in my P2P SIP/RTP VoIP application for almost a year now; it's been working wonderfully and I can't thank you enough. However, the manual makes no mention of the jitter buffer, nor does it (so

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

> > 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

Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple "hello world" dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

On May 3, 2006, at 9:54 PM, Jean-Marc Valin wrote: >> 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

On May 3, 2006, at 9:12 PM, Jean-Marc Valin wrote: >> 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

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

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

http://bugs.freedesktop.org/show_bug.cgi?id=24596 Summary: Video adapter #0 is jittery, while adapter #2 isn't. Product: xorg Version: unspecified Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at

> 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

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

Mike Taht wrote: > > > On 5/3/06, *Jean-Marc Valin* <Jean-Marc.Valin@usherbrooke.ca > <mailto: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

It seems that speex jitter buffer is tightly coupled with SPEEX codec [we have to give a speex decoder instance to JB]. It would be better if we could use it with any codec, like speex preprocessor and AEC. What are the any paper/theory/algorithms behind current ADAPTIVENESS of speex JB? Links to those algo/papers would help to understand better. -- Shantanu --- Thorvald Natvig