search for: 13kbps

...using ingress or u32. So this is how I did it: I called bellow script add_shaping DEV="eth0" tc qdisc add dev $DEV root handle 1: htb default 20 tc class add dev $DEV parent 1: classid 1:1 htb rate 200kbps ceil 200kbps tc class add dev $DEV parent 1:1 classid 1:15 htb rate 10kbps ceil 13kbps prio 3 tc class add dev $DEV parent 1:1 classid 1:20 htb rate 150kbps ceil 187kbps prio 2 tc qdisc add dev $DEV parent 1:15 handle 150: sfq perturb 10 tc qdisc add dev $DEV parent 1:20 handle 200: sfq perturb 10 U32="tc filter add dev $DEV parent 1:0 protocol ip u32" for computers in...

I've a question about the bandwidth consumed by IAX2/GSM. According to the wiki page, the GSM codec should run about 13 kilo-bits/sec for a voice encoding. However, watching gkrellm when I initiate a call to Digium, it looks like the channel is taking a consistent 5-6 kilo-bytes/sec. That's a lot more bandwidth than it should take. Is there perhaps a setting I have wrong somethere in

Hi... I have installed ip route 2 package on Linux kernel 2.4.25 I am using 2 tables: ################################### ebox:100.254~# ip route list table ALTER default via 192.168.100.253 dev br0 ebox:100.254~# ip route list table main 10.0.0.254 dev ppp0 proto kernel scope link src 10.0.0.1 192.168.100.0/24 dev br0 proto kernel scope link src 192.168.100.254 192.168.100.0/24 dev

Hi All, I've been playing with GSM and iLBC over low bandwidth connections (central Asterisk box with 2mbps, to ADSL users on 512/256) and both seem to perform well. Based upon what I've read in the archives and at voip-info.org iLBC should perform a little better if packets are lost, than compared to GSM. Do you find this to be true in practice, or is GSM just as robust? Whilst

...MB, 6kbps) Opus (--speech --set-ctl-int 4008=1103 --bitrate 56 --vbr --comp 5 --ignorelength - %d): (22:23, 9.3 MB, 74kbps) Opus (--speech --set-ctl-int 4008=1103 --bitrate 6 --vbr --comp 5 --ignorelength - %d): (22:23, 1.1 MB, 74kbps) Opus (--bitrate 12 --vbr --comp 5 --ignorelength - %d): (2.1MB, 13kbps) Regards, Ulrich >>> Mathias Buhr 30.10.2019, 14:39 >>> Hi Ulrich, I assume you've been using opusenc to encode the files. If that is the case you can try giving the encoder some more hints about your files: opusenc --speech --set-ctl-int 4008=1103 ... The latter should tell...

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

...tried outputting via WinAMP Disk Writer plugin to formats supported by the Windows audio codec manager, such as Windows Media v2 at various low bitrates in mono. All the results of compressed formats that worked at all came out with pretty severe artifacts. I also tried GSM 6.10 (8kHz sampling, 13kbps) which is acceptable on mobile phones and not bad here, with only slight tinny harshness. One of the big things with GSM and other telephony codecs is minimising latency (coding delay), but that didn't interest me, so I'm not surprised I did better with a different codec. I wasn't...

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa