similar to: RTP Packetization

To my knowledge, Asterisk's packetization rate is hard coded at 30ms. If I wanted to, where in the code could I go to change it to 20ms. Is there anything bad that might happen if I change it (asterisk related)? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/6088397a/attachment.htm

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

> 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

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?

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

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

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

Hello all, I'm trying to compile samba 3.0.20 on Freebsd 4.10 with ldap support. The ./configure script fails because it can't find ldap.h: <snip> checking whether to use AFS fake-kaserver... no checking whether to use DFS clear-text auth... no checking for LDAP support... auto checking ldap.h usability... no checking ldap.h presence... no checking for ldap.h... no checking lber.h

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

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a

Hello, *Background:* The RFC 5574 suggests the RTP payload format for the speex codec. The payload formation is straight forward; the encoded frames are to be concatenated one after another. Once we have appended desired number of frames, we have to pad the stream with 01111 sort of sequence to ensure that payload ends on a octet boundary. *Observation:* I am using the speex encoder at 2150 Kbps

Hi, I would like to use the package netalg made for Stefan Neubauer and Georg Dorffner. In this package there are functions similars with netlab, a package for Matlab. I searched for this package and didn't found it. This package had his name changed??? How I can download this package?? Best Regards André Barbosa Oliveira Student of Master in Economics at University Federal of Rio

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 3/21/23 18:28, Eric Blake wrote: > it is indeed a bug in busybox now that POSIX is moving towards > standardizing realpath, so I've filed it: > https://bugs.busybox.net/show_bug.cgi?id=15466 I've found another busybox bug. The "/bin/sh" utility is provided by busybox as well (via the usual symlinking). Per POSIX, if execvp(file, { argv[0], argv[1], ..., NULL })

Hi all We are about to send an updated version of the internet draft "RTP Payload Format for the Speex Codec" to the IETF AVT working group. Before submitting we would like your input, if you have any comments or input please send them to the mailing list. If we don't get any comments in 1 week (by 22. May 2007) we will go ahead and submit it to the IETF. Of course you can comment

You cannot concatenate bytes because Speex frames don't necessarily end on octet boundaries. You need to call the encoder multiple times on the same SpeexBits bitpacket. Jean-Marc Manish Jalan wrote: > Hello, > > _*Background:*_ > The RFC 5574 suggests the RTP payload format for the speex codec. The > payload formation is straight forward; the encoded frames are to be >

Hello Jean-Marc We really appreciate your input. If I understand it right, we should be calling the encoder on the same SpeexBits structure passing it a frame at a time to encode for as many times as the number of frames that we want to pack in the RTP payload. The output then obtained from from the encoder will have the necessary padding at the end without any separators between individual

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.

>> 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

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