similar to: Packetization Rate

Hi all, I need to be able to adjust packet sizes and found the patch at http://bugs.digium.com/view.php?id=5162 Thus, I checked out and compiled http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetization I added the line "packetization = 30" for one peer in my sip.conf and started asterisk with the "-I" switch for async RTP. That's all it takes

Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/b0cc470f/attachment.htm

Well, I don't know how to tune it more, it connects at about that rate in a mediocre rural landline. ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more of the necesary scarce bandwidth AND dropping sample info in each frame, thus making audio choppy and unclear. Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in

Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the "number of frames per RTP packet". How does this equate to packetization in ms?

I have an installation that connects in a [very] good day at 22kbps, but the normal is about 18kbps. I use de ILBC codec, and also change in iax.conf the trunkfreq = 20 to trunkfreq = 30 It works, you can understand well the other person, but don't expect miracles or an outstanding sound quality. > Dear Dan; > > Thanks alot for your kindly reply. > > Well, what u advise us

> Page 3: > > To be compliant with this specification, implementations MUST support > 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. > The sampling rate MUST be 8, 16 or 32 kHz. > > There is a type above after (narrowband), there is a " extra character. > > I don't understand what is the motivation to specify "SHOULD

Here's the story... Nortel system set to use g711 @ 30ms payload ... Asterisk box would need to communicate to that box @ 30 ms and another end point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 or 1.6.x... I've a preference for

Respected Sir, I want your help regarding an issue on asterisk. I hope my mail will not disturb your daily routine. My issue is I am connecting two asterisk over IIAX2/SIP trunk. I have successfully connected multiple server and every client from one server to call any other server's client. But problem is I want to use Speex@ 2.15kbs and also packetization time is 60ms but I can not

Hi all, I have spent some time searching, but I haven't found a way to prevent * from concatenating two frames into one IAX packet. I have a situation where I make an IAX GSM call to *, which transcodes to an iLBC SIP call. Every second voice packet the IAX client receives contains 2x 20ms frames, the other containing only one. I presume this is related to the mismatch of 20ms GSM vs

comment inline. On Wed, 16 May 2007, Jean-Marc Valin wrote: >> Page 3: >> >> To be compliant with this specification, implementations MUST support >> 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. >> The sampling rate MUST be 8, 16 or 32 kHz. >> >> There is a type above after (narrowband), there is a " extra

Hi Opus list, Please forgive me if this has been asked before. I find that Opus encoder created in mode OPUS_APPLICATION_AUDIO (as opposed to _VOIP) is using a lot of bits to encode silent periods of speech. This is relevant to a voip application for which good quality music is desirable, and in which I add a minimal comfort noise (order of few bits loud, e.g. MLS signal of amplitude 1 or 2)

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

Hi all, I''m running some tests with netem and I noticed some strange behaviour that looks like a bug: I''m pinging another machine and adding delay with netem. When I tell netem to give me a 10ms delay, it works fine. The problem is that when I ask for a 11ms delay, it gives me 20ms! It happens for any value between 11ms an 20ms, and it repeats for values over 20ms, now

I'm not an expert either, but I see people choosing iLBC over speex all the time with asterisk; partly it's because they have more market share in hardphones, and partly it's because of marketing and such. (another reason is that iLBC source is included in asterisk, and speex is only compiled in if you have the speex development stuff on your machine when you compile

Here my comments: Page 3: To be compliant with this specification, implementations MUST support 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. The sampling rate MUST be 8, 16 or 32 kHz. There is a type above after (narrowband), there is a " extra character. I don't understand what is the motivation to specify "SHOULD support 8 kbps

Hi, I''m trying to set up 15 different delay intervals for packets leaving on an interface, using netems hanging off of a 16-band prio. I''m having trouble adding anything to bands higher than 10. Here''s what I tried: tc qdisc add dev eth0 root handle 1: prio bands 16 \ priomap 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 I want all default traffic to go to

>> The main idea is that Speex supports many bit-rates, but for one reason >> or another, some modes may be left out in implementations (e.g. for RAM >> or network reasons). What we're saying here is that you should make an >> effoft to at least support (and offer) the 8 kbps mode to maximise >> compatibility. > > I understood this. But as you may know: the

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms

Jeff, RFC-5574 is standards-track: http://tools.ietf.org/html/rfc5574 so, while it's not an approved standard, it's more standardized than a lot of interoperable traffic on the internets these days. The RFC specifies packetization guidelines, which is basically that you put one or more frames in a packet, and then pad the rest with 0 bits until you have a while number of octets.

Hello Jean-Marc and all, I recently had a talk with somebody about CELP. He said, there is this federal standard 1016 (4.8kbps) with a reference implementation of the Department of Defense (only on Sun, unfortunately, if I got this right). This one is noticed in the manual already. He also said, since there is this implementation of the DoD, nobody would voluntarily re-implement CELP. If I read