similar to: sip jitter buffer

I've been trying to resolve some quality issues and I was hoping someone might be able to provide some insight. To give you an idea the calls are coming in via a SIP DID and sent out via an IAX2 connection. Latency to both the SIP equipment and IAX equipment are around 80ms with 0 packet loss accoridng to ping tests. The server is located in a data centre so bandwidth is not an issue. Most

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

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

Hi, I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is kind of a repost of one part of a previous question I have had. Peer Username Call ID Seq (Tx/Rx) Lag Jitter Format 213.137.73.178 xxxxxxxxxx 3705df0a5f7 00103/00000 00000ms 0000ms 4 1 active SIP channel(s) I see that there is 0ms Jitter set. How can I set a Jitter buffer for use with sip channels? I can't seem to find any documentation about this.

Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them before the end of the year to 1.4. Two of the main reasons that I would do this are Variable

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

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.

There is a theory that says that jitter buffers should not be used until the end of the voice path where jitter might be introduced. With that in mind, and in this scenario, the jitter buffers should reside at the ATA and media gateway; ATA (SIP UA) <> ASTERISK NATIVE BRIDGE <> MEDIA GATEWAY (SIP TO TDM) That raises a question about the Asterisk Native Bridge; Are the UDP RTP

> I'd like know about anyone using the current jitter buffer in Speex. I'm > planning on changing it to make it more general and I'd like some > feedback about how to make it better. Also, let me know if you're doing > anything serious with it and want to make sure I don't break your stuff. > > Basically, I want to make the jitter buffer easier to use with

Hi, Asterisk 1.4 Working (jitter buffers created as expected): ZAP -> SIP SIP -> ZAP Not working (no jitter buffers created): SIP -> chan_local (with /nj) -> ZAP SIP -> chan_local (with /j) -> ZAP SIP -> chan_local (with no flags) -> ZAP I have this in zapata.conf: jbenable=yes jbforce=no jbimpl=fixed jbmaxsize=300 Is there something I haven't tried that will make

Greetings list, Some time ago (probably about a year ago now) we disabled IAX jitter buffering on all our boxes because it was causing issues in a mixed 1.0 and 1.2 environment. One thing I've noticed over the last few months as more and more clients have moved from the 512k/1mb/2mb ADSL connections they were using onto "up to 8mb" connections is that whilst overall throughput is a

Hi all, is there a way to monitor SIP jitter buffer? I mean: in iax there are "iax2 show netstat" or "iax2 show channels" ...there is samething also for SIP protocol? Thanks alot in advance Lorenzo -- "Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non conta un cazzo, 1941 ... sono anche un autore"

What is SIP jitter buffer how can i test it ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Get easy, one-click access to your favorites. Make Yahoo! your homepage. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, I am keen to try out the SIP jitter buffer capability. I hear this was available if HEAD. I was wondering if a version of the latest STABLE with this additional feature was available some place.. Or is it simply best to use HEAD? Would some one be kind enough to point me in the right direction, Thanks, James

Thorvald Natvig wrote: >> I'd like know about anyone using the current jitter buffer in Speex. I'm >> planning on changing it to make it more general and I'd like some >> feedback about how to make it better. Also, let me know if you're doing >> anything serious with it and want to make sure I don't break your stuff. >> >> Basically, I want

Hello, The way you describe how the jitter buffer should be implemented makes me wonder: How does the jitter buffer works when there is no transmission? Let's say my "output" thread gets a speex frame from the jitter buffer every 20ms. What happen when there is no frame that arrived on the socket? No frames at all for a pretty long time (ie many seconds). This is my case because I

Hello all, I am currently investigating the JitterBuffer struct provided in the Speex library, and I am actually thinking about using it with two different codecs: namely, Speex-NB and AMR-NB. From looking at the code, it seems that JitterBuffer is capable of working for any codec. Both Speex-NB and AMR-NB (and probably also other narrowband codecs) produce 20 ms frames and the sampling frequency

> Hello, Hi :) First off, could you try to set your email client to break long lines before transmitting? In my (somewhat outdated) pine, the lines appear as VERY long lines when I try to reply, making it hard to read :) Minor detail though, I should probably fix pine. Some day. > The way you describe how the jitter buffer should be implemented makes me > wonder: How does the