similar to: VAD/CNG

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

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

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 transmission of VAD/CNG samples by specifying "outbound" in the communication channel only 1 bit instead of sending the whole VAD/CNG packet. I am doing the same for AMR

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

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

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

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 It is out of date, and doesn't support VAD nor silence detection.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

Hello. I am working on implementing RFC 5574 (RTP Payload Format for the Speex Codec) in the ffmpeg and have a question concerning it. It would be nice if somebody could answered it. * Chapter 4.1.1 Registration of Media Type Audio/Speex, subpart "Optional parameters" states these SDP optional parameters: vbr: variable bit-rate - either 'on', 'off', or 'vad'

Hello on Behalf Of idont know, Sangoma has a Media Gateway solution via SS7. They I believe are the only ones capable of connecting Asterisk via SS7. You may want to check them out. Heidi -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of idont know Sent: April 6, 2006 10:29 AM To: asterisk-biz@lists.digium.com

Hi All ;) I use VoipBuster with a IAX2 trunk between them and my 1.09 Asterisk. I use the G729 codec over my poor 64ko isdn line but when i putt a call on hold, le personn i call does't hear the music on hold (or just for some seconds) It seems that the VAD is enable on the IAX link. How can i disable it ? Is there something to add into my iax.conf file ? Thanks for the reply ;)

Just shoved my 7960 onto one of my clients networks. I'm talking from outside the office to them (they are using a GS BT101). I'm getting the following message repeatedly in the log, whenever someone talks. Jul 17 12:29:40 NOTICE[96402352]: frame.c:120 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Both phones are set to use the G729

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

Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 "Unsupported Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, My setup is SIP Call-->Asterisk-->VSP1 or VSP2 or VSP3 I'm experiencing an interconnect issue with one of the VSP's that seems to have to do with Asterisk not having any VAD control. The error is: NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The VSP has switched off silence suppression on their

All: Attached please find yet another RTP profile revision (v5). You can also find the document at: http://www.herlein.com/downloads/speex/docs/ Changes: - added vbr, cng, ebw, sr optional parameters to MIME - added vbr, cng, ebw a=fmtp options for SDP use - added required document attributes for submission to IETF and IANA (format and author contact information). Note that we

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.

Here is the setup: Phone -SIP G729-> AsteriskA -IAX G729-> AsteriskB -SIP G729-> Carrier The call completes but AsteriskA prints on the screen a ton of those "Dropping Frame of G729" messages starting about 5 seconds into the call: Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Jun

Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > > I've been using Speex in my voice-over-IP program on Win32, in > > wideband (16kHz) mode. I just starting using VBR recently and > > have run into something that might be a problem within Speex: > > Are you turning on DTX in addition to VBR? Also, what version are you > using. As of 1.0, DTX is

Hi Marc, Thanks for your quick response. So if VAD is enabled then VBR will be enable although it's a special VBR. How about take out the VAD code from the VBR and remove the code else? Lianghu On 10/24/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > 1) there are VAD related codes in the source file preprocess.c. > > Will VAD still work If I

As I understand it, there are two separate ways to get VAD information from Speex: 1) Using the encoder. 2) Using speex_preprocess(). I present the following observations from an application developer's perspective. They may be wrong, in which case I would appreciate corrections. - The two VAD systems are implemented differently. - speex_preprocess()'s VAD provides more accurate