search for: kpbs

...- Hash: SHA1 Hi, I was just measuring the bitrates of a couple of codecs via iax. I'm getting much higher numbers than expected, so maybe I'm doing something wrong? Measured with iptraf, values displayed are: codec: measured bitrate (bitrate according codec definition) gsm: 52 kbps (13 kpbs) alaw: 154 kbps (?) speex: 57 kpbs (24 kpbs) Seems a little high to me? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/huQT2TEAILET3McRAvEvAJ99hRxmGdCCqRGIpQSEMSwg6g5TBQCgnEFI zeYtRQE3H58KJ74...

jitter varioues from application to application and network to network. let me give you an instance from my own work. i am using speex in a voip application that has to work on gprs connection. the gprs gives just 9.6 kpbs up. hence, i have an upper limit of 6kpbs on the codec and not more than 5 udp packets per sec. that means, putting in 10 speex frames in every rtp packet. i need to jitter compensate for about 50 msec, but that necessarily means holding back 10 frames in the buffer before the next packet arriv...

Hi, I'm trying to use the jitter buffer feature that comes with Speex but I'm getting unexpected latency. I wrote a client application that does VOIP-like functions and without using jitter buffer, the end-to-end latency is around 250 ms (I'm using lowband 5.97 kpbs). However, when I tried to incorporate the jitter buffer feature, the latency would grow as time elapsed (up to a few seconds). I tried hacking the code by changing the SPEEX_JITTER_MAX_BUFFER_SIZE constant in jitter.c from 200 to 20 and the latency went down to around 300ms. Am I doing somethin...

Hi I currently have a 3072kbps line that I'm splitting in half for 5 of my phones. That's 307.2kbps +/- a couple of kpbs. What is the minimum kbps for a phone to maintain clarity and volume? Joshua

...ing into the ogg file itself ? Either use a field unused when using the quality setting, or store it in a tag. Current decoders won't be broken, newer decoders can take advantage of it. Player plugins could then use this to show in stead of bitrate, fx. "Q3" in stead of "112 kpbs". IIRC the cd player plugin does something like this in winamp - been a long time since i've used windows now though =) Anyway, what do you think ? Could this be done ? -- Trick __________________________________________ Linux User #229006 * http://counter.li.org "There is no ma...

...a date or some other planning when all those already-in-the-format-but-not-yet-implemented features will be there. I don't want to end up with a situation saying 'yes; it now can do stereo; but the next version will have joint-stereo and allow a current 128 Kbps stream to be put into a 80 Kpbs stream'; when that 'next version' is not due for more then a year. I've never liked 'vaporware'; and -hence- I don't want to advocate it myself. ;-) Cheerio! Kr. Bonne. -- KB905-RIPE Belgacom IP networking (c=be,a=rtt,p=belgacomg...

...tton smartnode 4118/js/eiu fxs gateway with asterisk? We we're able to get the unit to register with asterisk, but when trying to place a call, no codec was compatible, even though I had all of the following enabled on the patton ... # G.711 A-Law/?-Law (64kbps) # G.726 (ADPCM 40, 32, 24, 16 kpbs) # G.723.1 (5.3 or 6.3 kbps) # G.729ab (8kbps) the link to this product is : http://commerce.patton.com/pe_products.asp?category=51&MiDAS_SessionID=e41363efa86e409caf79ab1fd9b32e49 ? thanks for any help, Michael Lyszczek

...when headers are incomplete, and gives junk info for the stream length and nan for the bitrate * doesn't say anything if I put 1k of random data at the end of a truncated stream, only 10k (then it says "Hole in data found.") * output wishlist: - bitrate name is inconsistant: kb/s or kpbs - some section headers in the output have ':', others '...' - some sections of data are indented, some not - printf-style formatted output - line up the default output My suggested output example, including a few other minor changes I didn't note above: --cut-- Found logic...

...esday, November 16, 2004 8:25 PM To: speex Subject: [Speex-dev] jitter buffer jitter varioues from application to application and network to network. let me give you an instance from my own work. i am using speex in a voip application that has to work on gprs connection. the gprs gives just 9.6 kpbs up. hence, i have an upper limit of 6kpbs on the codec and not more than 5 udp packets per sec. that means, putting in 10 speex frames in every rtp packet. i need to jitter compensate for about 50 msec, but that necessarily means holding back 10 frames in the buffer before the next packet arriv...

...mbit/s on a bonded dual gigabit nic (probably, with a bad bonding mode configured) fio returns about 50000kB/s, that are 400000 kbps. As I'm using replica 3, the host has to write to 3 storage server, thus: 400000*3 = 1200000 kbps If I understood properly, i'm able to reach about 1200000 kpbs on network side, with sequential writes, right ? Why a simple "dd" will return only 30MB/s ?

...esday, November 16, 2004 8:25 PM To: speex Subject: [Speex-dev] jitter buffer jitter varioues from application to application and network to network. let me give you an instance from my own work. i am using speex in a voip application that has to work on gprs connection. the gprs gives just 9.6 kpbs up. hence, i have an upper limit of 6kpbs on the codec and not more than 5 udp packets per sec. that means, putting in 10 speex frames in every rtp packet. i need to jitter compensate for about 50 msec, but that necessarily means holding back 10 frames in the buffer before the next packet arriv...

I have put a box together using 2 X100P and 2 TDM400 4port cards. Using the simple setup that Martin posted a few days ago, I have asterisk almost up and running. /etc/zaptel.conf fxsks=1-2 fxoks=3-10 loadzone=nz defaultzone=nz (I have added NZ tone information to zaptel and Asterisk - I'll submit a patch soon). /etc/asterisk/zapata.conf [channels] context=incoming signalling=fxs_ks

Hello folks Anyone know if this would be enough to decode oggs? AMD Elan SC520 133 MHz If not, what seems to be the lower limit required for decoding oggs? Many thanks, Kerry. <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only

Can anyone share their experience on the maximum number of calls a given asterisk box/asterisk software can handle? I see the asterisk business edition can handle up to 240 simultaneously with appropriate licensing, but that doesn't seem to be many at all. For now, I plan to use the stable open source versions - would it be reasonable to say that it is more of hardware limitation on the

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's