similar to: size of speex packets when VAD/CNG is enabled

you can use the marker bit on the RTP. - farhan On Tue, Aug 19, 2008 at 12:04 PM, Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote: > Hi all, > > supposing to have speex working at 3.95, when VAD/CNG is employed, which > is the size of those "silence" samples respect to voice samples? > > I am wondering if it could be possible to lower even the

Dear all, i am currently evaluating which codec to be used for a "very narrowband" voice communication system (using either VoIP either GSM CSD) on mobile devices running Windows Mobile and Symbian OS. I had experience in using AMR-NB 4.75 with VAD on Symbian OS and all worked very well. However, AMR it's not royalty free and if i need to use it on Windows Mobile i will need

Hi List, i've set up a cisco 7912 for my asterisk box. I've had problems with VAD and CNG. After googling a bit, i've found an article about asterisk not supporting these two protocols, therefore it's better to turn them off. Since then i did not found answer to my two questions, maybe somebody here could help me: a) am i even able to turn off vad/cng on cisco 7912? SIP

It would be very nice to know about some good success without too much cpu issues on Nokia devices, i tried pjsip.org project on S60 FP1 E65 +200mhz ARM cpu with speex but don't had much luck in using it without having 100% cpu usage :( Please let us to know about your testing E65 CPU: http://www.nokia-tuning.net/index.php?s=processor Fabio Pietrosanti Jordan Dimov wrote: > Thank

hi all, I can claim that our hotfoon service has run millions of mobile voip minutes using speex on mobile and desktops. it works! earlier you needed more powerful cpus (until the integer port), however now the cpus are much faster and the code complexity is much lower. my only regret is of not being able to write some demo apps for speex on win mob. I will surely deliver it in june to settle this

Quoting "Fabio Pietrosanti (naif)" <lists at infosecurity.ch>: > Speex it's too cpu expensive for general S60 usage, it would require a > lot of ASM optimization. Did a quick search and saw ARM CPUs with speeds above 100 MHz. That should actually be enough for Speex, at least for narrowband. > If you are using CSD, like for a secure telephony solution >

hello all, At my workplace, our colleague Mukesh has just completed a port of speex 1.2.1 to Palm OS. He had to write a couple of mathlib functions, but the trunk remains unchanged. It works quite well, I was wondering if there will be any interest in this port in the distribution. - farhan

Is speex at this stage a good codec to use for real-time voice transmission over a direct CSD connection between two phones? Does anyone have any experience with using Speex on Nokia phones? If speex is not a good solution for this task, what would you recommend? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm losing call files in /var/spool/asterisk/outgoing because * isn't able to detect the busy signal. The call file looks like this: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Using the "|caller" parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable

I know that asterisk does not support VAD. I poked around and saw some reference to asterisk supporting CNG. Will CNG work without VAD ? If yes is there any way to set this on asterisk 1.2.X for SIP using G729 ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a question regarding incoming fax to local file (on the Asterisk server). While the fax is received properly (I have the tiff file generated as expected) I get the warning 'FAX CNG detected but no fax extension' on the consol. If the fax is received ok then what 'fax extension' does it expect and what should I do there? My Setup: Sender -> Public PSTN ->

Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last

Jean-Marc Valin wrote: >>In particular, (I'm not really sure, because I don't thorougly >>understand it yet) I don't think your jitterbuffer handles: >> >>DTX: discontinuous transmission. >> >> > >That is dealt with by the codec, at least for Speex. When it stops >receiving packets, it already knows whether it's in DTX/CNG mode.

How to disable silence suppression (or Voice activity detection - VAD) on Cisco 7905 phone? On Cisco 7940 I use "enable_vad: 0", but I can't find anything similar for 7905. -- Tomislav Parcina tparcina#lama.hr

Dear list, I need to convert this data.frame > names(codesM) [1] "key" "AMR.pa1.M" "AMR.pa2.M" "AMR.pa3.M" "AMR.pa4.M" [6] "AMR.pa5.M" "AMR.pa6.M" "AMR.pa7.M" "AMR.pa8.M" "AMR.pa9.M" [11] "AMR.pa10.M" "AMR.ta1.M" "AMR.ta2.M" "AMR.ta3.M"

I have difficulties to understand how to use Silk GNC. I have enabled GNC (and VAD) something like this: param.setting.vad = 1; param.setting.cng = 1; param.setting.plc = 1; param.setting.penh = 1; status = pjmedia_codec_opus_set_default_param(&config, &param); I used debugger and printf to see that code executes in Opus CNG.c silk_CNG function noise generation loop. But still I cannot

Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. 2.) faxdetect=incoming will, upon detection of a CNG tone,

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and "b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see