similar to: Speex: Jitter and silence suppression

OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem?

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

After some more analysis of my "dropped fragment" problem, things look like this: Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

Hello, I have several customers describing something like annoying "silence suppression". So I did some tests and I can confirm. After disabling echo cancellation in zapata.conf the "silence suppression" effect is gone, but there is a little echo of course. I do not have this problem on other boxes where I am using oslec as an echo canceller. All my calls are SIP to ZAP (E1).

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

Hi Friends, Currently i am using the asterisk 1.4.x version. In that i want to enable to silence suppression in the SIP calls. Please tell me the configuration changes to be done. Thanks in advance, balasam. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090106/dde500d5/attachment.htm

Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that

Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman -------------- next part -------------- An HTML attachment was scrubbed... URL:

I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

> 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. > clock skew: (see discussion, though) Clock skew is one of the main

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

Hello, The way you describe how the jitter buffer should be implemented makes me wonder: How does the jitter buffer works when there is no transmission? Let's say my "output" thread gets a speex frame from the jitter buffer every 20ms. What happen when there is no frame that arrived on the socket? No frames at all for a pretty long time (ie many seconds). This is my case because I

> Thank you for a very good explanation which shed light on some of the > questions that I had after reading the source code. > > Reading your text however, I wonder if I'm perhaps missing an important > point on the proper use of the jitter buffer: > > ... >> Now, clearly, if early_ratio is high and late_ratio is very >> low, the buffer is buffering more than

>> Err, unless I'm totally wrong, there are a few race conditions. >> >> Assume the buffer is full of packets newer than the current pointer, and >> one that is at the current pointer. >> >> get and put start at the same time. >> >> get will find the correct buffer index. Now, just after it finds it's >> index, assume we switch to the

Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send

Hi Zachary, Zachary Schneirov a ?crit : > The speex_jitter_buffer.c wrapper seems to maintain (buffer?) one > packet-frame ("current_packet") in addition to the packets already > tracked by the JitterBuffer itself. Why is this necessary? That's in case there's more than one frame per packet. That way, it finds the packet, decodes the first frame it contains, and

> OK, I'm actually about ready to start working on this now. > > If people in the speex community are interested in working with me on > this, I can probably start with the speex buffer, but I imagine > there's going to be a lot more work needed to get this where I'd like > it to go. And where would you like it to go? ;-) > At the API level, It seems pretty easy

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.