Re: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
Brian Cuthie wrote: 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 don't emanate from a SIP phone here at my location. Asterisk SIP does not support silence suppression. In fact, using Silence suppression on an inbound RTP stream will lead to problems, since Asterisk takes timing from inbound RTP streams. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Brian Cuthie wrote: 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 don't emanate from a SIP phone here at my location. Asterisk SIP does not support silence suppression. In fact, using Silence suppression on an inbound RTP stream will lead to problems, since Asterisk takes timing from inbound RTP streams. Yeah, funny thing is I saw this problem just last night while messing around with music on hold. I had VAD on the SIP phone on and the MOH would stop unless I talked. I thought it was quite weird when it happened; now it makes sense. I've heard that Asterisk derives its timing in strange ways, but I've been wondering why it doesn't use the machine's clock (real-time interrupt, not wall-clock). -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
At 8:34 AM -0400 on 4/5/04, Brian Cuthie wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Brian Cuthie wrote: 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 don't emanate from a SIP phone here at my location. Asterisk SIP does not support silence suppression. In fact, using Silence suppression on an inbound RTP stream will lead to problems, since Asterisk takes timing from inbound RTP streams. Yeah, funny thing is I saw this problem just last night while messing around with music on hold. I had VAD on the SIP phone on and the MOH would stop unless I talked. I thought it was quite weird when it happened; now it makes sense. I've heard that Asterisk derives its timing in strange ways, but I've been wondering why it doesn't use the machine's clock (real-time interrupt, not wall-clock). -brian Interestingly enough, Mark and I talked about this problem very briefly at dinner the other night. My recollection is that he seemed to think that taking timing from a Zap driver would be feasible, but there were many other things to do ahead of time. Perhaps others can program this or encourage it's development. Personally, I think VAD is a great service, as well as comfort noise generation to disguise when VAD is working. I'll always encourage methods that reduce bandwidth. Most major developers on Asterisk consider these technologies of low concern since their bandwidth is unlimited, as they typically sit in a co-lo somewhere (as many programmers of * are providers of service, not consumers.) The reality for most end users is that they are on very restricted pipes that are delivered via a WAN technology (especially for outbound, if you consider residential) and being able to put more customers into expensive bitstreams makes a lot of financial sense. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
Personally, I think VAD is a great service, as well as comfort noise generation to disguise when VAD is working. I'll always encourage methods that reduce bandwidth. Most major developers on Asterisk consider these technologies of low concern since their bandwidth is unlimited, as they typically sit in a co-lo somewhere (as many programmers of * are providers of service, not consumers.) The reality for most end users is that they are on very restricted pipes that are delivered via a WAN technology (especially for outbound, if you consider residential) and being able to put more customers into expensive bitstreams makes a lot of financial sense. I agree fully. We need to implement a good timer in the SIP channel, both for VAD (but that's really in RTP, isn't it?) and for general SIP timers according to the RFC. Last week I also learned that DSL in the US is not as fat as DSL in general is over here. Anything below 384 upstream is nothing you can sell in Sweden :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
Title: Silence suppression on SIP calls generated from Asterisk? 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 don't emanate from a SIP phone here at my location. Thanks -brian
