RE: [Asterisk-Users] Calculating required bandwidth
Hi, > >> A T1 set up for voice carries 24 conversations on a circuit that is > >> 1.544 megabits/second. Right? > > > > Yes and no. If the T1 is channelized, then yes. If it's a PRI > > circuit, then it has only 23 channels to carry voice, as the 24th > > channel is used for the D-channel (signalling channel). > > Only if you're in the US. We have 30 + 1 :-) Are you sure? As far as I know, E1 is 30 + 2, not 1... Best Regards, -- Durval Menezes (durval AT tmp DOT com DOT br, http://www.tmp.com.br/) ___ 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] Calculating required bandwidth
Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit, then it has only 23 channels to carry voice, as the 24th channel is used for the D-channel (signalling channel). Only if you're in the US. We have 30 + 1 :-) E1 == 2048kbps == 32 channels, giving 30 B + 1 D + 1 for timing roy ___ 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] Calculating required bandwidth
[EMAIL PROTECTED] wrote: > [EMAIL PROTECTED] wrote: > >> [EMAIL PROTECTED] wrote: >>> I was posed this question: >>> >>> A T1 set up for voice carries 24 conversations on a circuit that is >>> 1.544 megabits/second. Right? >> >> Yes and no. If the T1 is channelized, then yes. If it's a PRI >> circuit, then it has only 23 channels to carry voice, as the 24th >> channel is used for the D-channel (signalling channel). > > Only if you're in the US. We have 30 + 1 :-) Nope. I'm in Canada. And what you are referring to is an E1, not a T1. ___ 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] Calculating required bandwidth
[EMAIL PROTECTED] wrote: > [EMAIL PROTECTED] wrote: >> I was posed this question: >> >> A T1 set up for voice carries 24 conversations on a circuit that is >> 1.544 megabits/second. Right? > > Yes and no. If the T1 is channelized, then yes. If it's a PRI > circuit, then it has only 23 channels to carry voice, as the 24th > channel is used for the D-channel (signalling channel). Only if you're in the US. We have 30 + 1 :-) -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] Calculating required bandwidth
--On Thursday, December 16, 2004 9:45 AM -0800 Ed Greenberg <[EMAIL PROTECTED]> wrote: I was posed this question: I've learned a ton, in the discussion that followed this question. Thanks, all. ___ 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] Calculating required bandwidth
[EMAIL PROTECTED] wrote: > http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption > > A T1 set up for voice uses G711 at 64Kbps which at 1.544Mbps > equals 24 channels. This is also known as a PRI. No, that is incorrect. PRI is a type of ISDN service that is typically carried across one (or more) T1s. The PRI-ISDN service uses one of the 24 channels in the T1 as a signalling channel (or D-channel), the other 23 channels become the bearer-channels (or B-channels). PRI is sometimes referred to as 23B+D, but in reality can be built far more creatively than that. Many large PBXs will have a single PRI circuit that is delivered on two T1s, with a primary and backup D-channel in each T1 (46B+2D). If you wanted, you could save the backup D-channel and have a 47B+D (bad idea, but technically possible). It is even possible to have a D-channel running across a serial link, completely out of the T1 altogether; although I haven't seen one of these in a long time . . . To summarize: A T1 is a *circuit*, that can be used to carry all kinds of different services. PRI is one kind of *service* that is commonly carried over a T1. Cheers, Jim. > > -Matthew > > - Original Message - > From: "Ed Greenberg" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Thursday, December 16, 2004 11:45 AM > Subject: [Asterisk-Users] Calculating required bandwidth > > >> I was posed this question: >> >> A T1 set up for voice carries 24 conversations on a circuit that is >> 1.544 megabits/second. Right? >> >> Well, if you set that T1 up to carry data and run a link between two >> IP networks over it, how many SIP conversations could it be expected >> to > carry? >> How about IAX? >> >> How would one extend this calculation to varying bandwidth circuits >> and various VOIP protocols (MGCP, SCCP and H323 come to mind)? >> >> Rather than asking for a full education here, can somebody point me >> at a suitable practical reference? Of course, if somebody wants to >> actually > post >> the answer that'd be fine too :) >> >> THanks, >> >> >> >> ___ >> 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 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 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] Calculating required bandwidth
[EMAIL PROTECTED] wrote: > --On Thursday, December 16, 2004 3:59 PM -0500 Jim Van Meggelen > <[EMAIL PROTECTED]> wrote: > >> I've always found Newton's Telecom Dictionary to be a great >> reference. It's not too technical, packed with humour, and very >> comprehensive. > > I have a very old copy of this, so went off to Amazon to see about a > new one. I discovered that that a 2005 edition (21st edition) > will be available > in February, so I'm going to wait. That reminds me of last January, when I said to myself "The 2004 version is coing out in a few weeks, I'll wait". I just picked mine up last week :-) Regards, Jim. ___ 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] Calculating required bandwidth
--On Thursday, December 16, 2004 3:59 PM -0500 Jim Van Meggelen <[EMAIL PROTECTED]> wrote: I've always found Newton's Telecom Dictionary to be a great reference. It's not too technical, packed with humour, and very comprehensive. I have a very old copy of this, so went off to Amazon to see about a new one. I discovered that that a 2005 edition (21st edition) will be available in February, so I'm going to wait. ___ 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] Calculating required bandwidth
On December 16, 2004 03:49 pm, Race Vanderdecken wrote: > Also remember that a telephone conversation is 2/3's silence. ( I speak, > silence, then you speak. See the book at bought on Amazon 4 years ago > but can't remember the name of the book.)IP only sends the data when > there is noise versus the T1 which is a constant TDM stream. So I Incorrect. Only when VAD is active is silence supression used. Asterisk does not currently support VAD in any form since the RTP stream is used as a clock source. > predict in testing with good VoIP equipment you can get more then 24 > G.711 calls per T1. So take that and comment. You should be able to get > more VoIP calls, my prediction is 40 G.711 well behaved calls with > silence suppression per T1. Why else would the Baby Bells move to VoIP? Because they'll transcode to GSM (minimum) and call it toll quality? :-) -A. ___ 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] Calculating required bandwidth
[EMAIL PROTECTED] wrote: > I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit that is > 1.544 megabits/second. Right? Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit, then it has only 23 channels to carry voice, as the 24th channel is used for the D-channel (signalling channel). PRI is superior, because it offers far more flexible use of the circuit, and provides far more information (like CallerID). > Well, if you set that T1 up to carry data and run a link between two > IP networks over it, how many SIP conversations could it be expected > to carry? How about IAX? Interesting question. I'll tell you this, it won't have so much to do with the 24 channels as it will with how efficiently the circuit is used. When you run data on a T1, all of the pipe is treated as one big channel by the upper layers. The 24 timeslots are all still there, but the network doesn't have any knowledge of them. > How would one extend this calculation to varying bandwidth circuits > and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Each network layer (think of the OSI model) will add overhead, so the calculation has to take into account how the data (in this case, the voice packets) is encapsulated at each layer. Of the protocols, IAX would probably utilize the circuit most efficiently, due to it's trunking. Naturally, the codec you use will be another key factor. > Rather than asking for a full education here, can somebody point me > at a suitable practical reference? Of course, if somebody wants to > actually post the answer that'd be fine too :) I've always found Newton's Telecom Dictionary to be a great reference. It's not too technical, packed with humour, and very comprehensive. ___ 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] Calculating required bandwidth
Thank you Peasants, In general the original question was answered. I am software guy, if the network slobs can't fit all the data in the pipe that is not my problem. The basic idea in the answer was that you can get more calls by using compression; much like the automobile manufacture's gas mileage may vary. Also remember that a telephone conversation is 2/3's silence. ( I speak, silence, then you speak. See the book at bought on Amazon 4 years ago but can't remember the name of the book.)IP only sends the data when there is noise versus the T1 which is a constant TDM stream. So I predict in testing with good VoIP equipment you can get more then 24 G.711 calls per T1. So take that and comment. You should be able to get more VoIP calls, my prediction is 40 G.711 well behaved calls with silence suppression per T1. Why else would the Baby Bells move to VoIP? But it is nice to know there are some intelligent folks monitoring the list, thank you. Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: 16 December 2004 14:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Calculating required bandwidth On December 16, 2004 01:52 pm, Race Vanderdecken wrote: > The quick tyrannical answer, And wrong -- I am taking the time to correct it not so much to slam you but more for list posterity -- just because the codec rate is 64kbps doesn't mean that's what's actually on the wire, even if you ignore signalling. > Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 each T1 has 24 channels of 8 bit data plus one frame bit. 24*8+1 = 193 bits per T1 frame. Frames are sent 8000 per second. 8000*193 = 1544000 bits per second. There's your T1 raw rate. You can't use that frame bit for yourself so 24*8*8000 = 1536000 bits per second. That's your T1 data rate; that's what you can actually use. Now. Running IP on a T1 you have certain overheads. UDP frame overhead is 4 bytes, plus your TCP overhead of 12 bytes, for a total of 16 bytes (128 bits). G.711 is 64kbps data rate, but Asterisk sends only 20ms per packet in an attempt to balance data throughput and effect of lost packets. so 64kbps / 50 is 1280 bits of audio per packet, plus 128 bits of overhead for 1408 bits per packet. 50 of these per second of audio gives you 70400bps for one second of G.711 VOIP audio. so now take your T1 data rate of 1536000bps and divide your audio rate into it for an answer of 21 channels of G.711 VOIP audio. Now that was straight UDP audio -- there was no signalling overhead and it wasn't SIP RTP. RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it is actually costlier: I'll spare you all the calculations but it's 20 channels of SIP G.711 audio per T1, likely with enough room for signalling. :-) Regards, Andrew "the tyrant's tyrant" Kohlsmith ___ 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 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] Calculating required bandwidth
You can encapsulate it as ppp, still some overhead, but less I think than HDLC. Ed - Original Message - From: "Kevin P. Fleming" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, December 16, 2004 1:36 PM Subject: Re: [Asterisk-Users] Calculating required bandwidth > Andrew Kohlsmith wrote: > > > RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it > > is actually costlier: I'll spare you all the calculations but it's 20 > > channels of SIP G.711 audio per T1, likely with enough room for > > signalling. :-) > > And you can't run straight IP over a T1 circuit either; it's usually > framed in HDLC frames. There's a little more overhead for you > > 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 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] Calculating required bandwidth
A TDM channel (POTS/PRI) consists of 64k of real-time data, VoIP with no compression (g.711) consists of 64k plus IP protocol overhead for a total bandwidth or 80 to 90k required per uncompressed channel. So a IP T1 carrying VoIP without compression has lower capacity that a Voice T1. A t1 for voice typically carries 23 b channels and 1 d channel, so 23 conversations not 24. If you use compression on the VoIP traffic you gain capacity, but loose CPU performance as the RTP data stream has to be transcoded by *. If compression is used, and the box has the CPU power, significantly more than 23 is the answer, probably limited more by then number that your * can setup, transcode, and tear down. The exact answer depends on your use and can only be determined through testing. Uncompressed the answer is probably closer to 15 to 18 RTP streams across a dedicate T1 IP link. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Ed Greenberg > Sent: Thursday, December 16, 2004 10:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Calculating required bandwidth > > I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit > that is 1.544 megabits/second. Right? > > Well, if you set that T1 up to carry data and run a link > between two IP networks over it, how many SIP conversations > could it be expected to carry? > How about IAX? > > How would one extend this calculation to varying bandwidth > circuits and various VOIP protocols (MGCP, SCCP and H323 come > to mind)? > > Rather than asking for a full education here, can somebody > point me at a suitable practical reference? Of course, if > somebody wants to actually post the answer that'd be fine too :) > > THanks, > > > > ___ > 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 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] Calculating required bandwidth
On December 16, 2004 02:36 pm, Kevin P. Fleming wrote: > And you can't run straight IP over a T1 circuit either; it's usually > framed in HDLC frames. There's a little more overhead for you Augh you are absolutely correct... See even the tyrant's tyrant screws it up. :-) -A. ___ 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] Calculating required bandwidth
Andrew Kohlsmith wrote: RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it is actually costlier: I'll spare you all the calculations but it's 20 channels of SIP G.711 audio per T1, likely with enough room for signalling. :-) And you can't run straight IP over a T1 circuit either; it's usually framed in HDLC frames. There's a little more overhead for you ___ 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] Calculating required bandwidth
On December 16, 2004 02:00 pm, Damon Estep wrote: > A TDM channel (POTS/PRI) consists of 64k of real-time data, VoIP with no > compression (g.711) consists of 64k plus IP protocol overhead for a > total bandwidth or 80 to 90k required per uncompressed channel. So a IP > T1 carrying VoIP without compression has lower capacity that a Voice T1. > A t1 for voice typically carries 23 b channels and 1 d channel, so 23 > conversations not 24. Voice channelized T1 (also known as CAS T1 in Canada) is 24 channels. PRI is (simplified explanation) out of band signalling on a DS1, but uses 1 channel for signalling (it's out of band now, so it has to go somewhere) so you get 23 channels of voice and 1 for the signalling. Data T1 carrying VOIP traffic will be able to handle about 21 channels of G.711 RTP audio due to RTP and IP overhead, and does not include SIP/H.323 signalling, although the signalling overhead should be able to fit in the remainder of the T1. > Uncompressed the answer is probably closer to 15 to 18 RTP streams > across a dedicate T1 IP link. That few? I would be surprised if the signalling overhead is that enormous. With 21 channels of RTP G.711 audio I have 64kbps of bandwidth available, at least according to my calculations. -A. ___ 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] Calculating required bandwidth
On December 16, 2004 02:00 pm, Matthew Boehm wrote: > http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption > > A T1 set up for voice uses G711 at 64Kbps which at 1.544Mbps equals 24 > channels. This is also known as a PRI. No it's not also known as a PRI; PRI is a very specific (out of band) signalling on top of physical T1 (DS1) lines. -A. ___ 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] Calculating required bandwidth
On December 16, 2004 01:52 pm, Race Vanderdecken wrote: > The quick tyrannical answer, And wrong -- I am taking the time to correct it not so much to slam you but more for list posterity -- just because the codec rate is 64kbps doesn't mean that's what's actually on the wire, even if you ignore signalling. > Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 each T1 has 24 channels of 8 bit data plus one frame bit. 24*8+1 = 193 bits per T1 frame. Frames are sent 8000 per second. 8000*193 = 1544000 bits per second. There's your T1 raw rate. You can't use that frame bit for yourself so 24*8*8000 = 1536000 bits per second. That's your T1 data rate; that's what you can actually use. Now. Running IP on a T1 you have certain overheads. UDP frame overhead is 4 bytes, plus your TCP overhead of 12 bytes, for a total of 16 bytes (128 bits). G.711 is 64kbps data rate, but Asterisk sends only 20ms per packet in an attempt to balance data throughput and effect of lost packets. so 64kbps / 50 is 1280 bits of audio per packet, plus 128 bits of overhead for 1408 bits per packet. 50 of these per second of audio gives you 70400bps for one second of G.711 VOIP audio. so now take your T1 data rate of 1536000bps and divide your audio rate into it for an answer of 21 channels of G.711 VOIP audio. Now that was straight UDP audio -- there was no signalling overhead and it wasn't SIP RTP. RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it is actually costlier: I'll spare you all the calculations but it's 20 channels of SIP G.711 audio per T1, likely with enough room for signalling. :-) Regards, Andrew "the tyrant's tyrant" Kohlsmith ___ 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] Calculating required bandwidth
"So if you use G.711 codec then you will be able run 24 SIP conversations on a T1." Not true, the RTP stream is 64k, there is also IP packet and VoIP protocol overhead to deal with. If you try to dedicate less than 80k+ per g.711 stream you will have trouble and you also have IP data on the T1 you are really looking for trouble. Priority queuing will help, but only if you have enough bandwidth to transfer all of your RTP and data packets. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Race Vanderdecken > Sent: Thursday, December 16, 2004 11:53 AM > To: 'Ed Greenberg'; 'Asterisk Users Mailing List - > Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Calculating required bandwidth > > The quick tyrannical answer, > > Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 > > G.711 CODEC is used on the T1 Channels. > > So if you use G.711 codec then you will be able run 24 SIP > conversations on a T1. > > SIP is not a codec, SIP is a call control protocol. SIP does > the work of connecting two endpoints together. MGCP, SCCP and > H323 are also just call control protocols. SIP has low > overhead while H323 has lots of features. The overhead is so > small that it won't really figure in the CODEC calculations. > > The RTP protocol is responsible for moving the voice data > from point to point. But I digress. > > When you look at the CODECs each compresses the voice data > differently. It is this compression that gives you your > number of "phone calls" on a T1. > > As per -- > http://www.vocal.com/data_sheets/full/code_source_voip_g723.html > > Calls per T1 | Codec explanation > 289 or 240|*G.723 (often referred to as G.723.1) - 5 1/3k > and 6.4k bps ACELP/MP-MLQ > 193 |*G.729 - 8k bps CS-ACELP *G.729A - reduced > complexity versionof G.729 - fewer MIPS > at the expense of reduced perceived signal quality > 118 |*GSM 06.10 - 13k bps RPE-LTP > 96|*G.728 - 16k bps LD-CELP > 96 to 38 |*G.726 - 16k, 24k, 32k and 40k bps ADPCM - > normally not used in Voice-over-IP applications > 48|*G.721 - 32k bps ADPCM - normally not used in > Voice-over-IP applications > 24|*G.711 - 64k bps PCM (A-Law or m-Law format) > > The above table shows one of the reasons G.729 is popular in > that you can get 192 calls per T1 with fair quality. > > Remember time is money; the tighter the compression the more > time it takes to compress/decompress and therefore the more > money in silicon it takes to do the compressions on the fly. > Smaller call "channel/bandwidth" means more hardware > horsepower to compress and decompress the voice on the call. > > Race "The Tyrant" Van der Decken > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Ed Greenberg > Sent: 16 December 2004 12:45 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Calculating required bandwidth > > I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit > that is 1.544 megabits/second. Right? > > Well, if you set that T1 up to carry data and run a link > between two IP networks over it, how many SIP conversations > could it be expected to carry? > How about IAX? > > How would one extend this calculation to varying bandwidth > circuits and various VOIP protocols (MGCP, SCCP and H323 come > to mind)? > > Rather than asking for a full education here, can somebody > point me at a suitable practical reference? Of course, if > somebody wants to actually post the answer that'd be fine too :) > > THanks, > > > > ___ > 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 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 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] Calculating required bandwidth
http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption A T1 set up for voice uses G711 at 64Kbps which at 1.544Mbps equals 24 channels. This is also known as a PRI. -Matthew - Original Message - From: "Ed Greenberg" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, December 16, 2004 11:45 AM Subject: [Asterisk-Users] Calculating required bandwidth > I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit that is 1.544 > megabits/second. Right? > > Well, if you set that T1 up to carry data and run a link between two IP > networks over it, how many SIP conversations could it be expected to carry? > How about IAX? > > How would one extend this calculation to varying bandwidth circuits and > various VOIP protocols (MGCP, SCCP and H323 come to mind)? > > Rather than asking for a full education here, can somebody point me at a > suitable practical reference? Of course, if somebody wants to actually post > the answer that'd be fine too :) > > THanks, > > > > ___ > 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 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] Calculating required bandwidth
The quick tyrannical answer, Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 G.711 CODEC is used on the T1 Channels. So if you use G.711 codec then you will be able run 24 SIP conversations on a T1. SIP is not a codec, SIP is a call control protocol. SIP does the work of connecting two endpoints together. MGCP, SCCP and H323 are also just call control protocols. SIP has low overhead while H323 has lots of features. The overhead is so small that it won't really figure in the CODEC calculations. The RTP protocol is responsible for moving the voice data from point to point. But I digress. When you look at the CODECs each compresses the voice data differently. It is this compression that gives you your number of "phone calls" on a T1. As per -- http://www.vocal.com/data_sheets/full/code_source_voip_g723.html Calls per T1 | Codec explanation 289 or 240 |âG.723 (often referred to as G.723.1) - 5 1/3k and 6.4k bps ACELP/MP-MLQ 193 |âG.729 - 8k bps CS-ACELP âG.729A - reduced complexity version of G.729 - fewer MIPS at the expense of reduced perceived signal quality 118 |âGSM 06.10 - 13k bps RPE-LTP 96 |âG.728 - 16k bps LD-CELP 96 to 38|âG.726 - 16k, 24k, 32k and 40k bps ADPCM - normally not used in Voice-over-IP applications 48 |âG.721 - 32k bps ADPCM - normally not used in Voice-over-IP applications 24 |âG.711 - 64k bps PCM (A-Law or ï-Law format) The above table shows one of the reasons G.729 is popular in that you can get 192 calls per T1 with fair quality. Remember time is money; the tighter the compression the more time it takes to compress/decompress and therefore the more money in silicon it takes to do the compressions on the fly. Smaller call "channel/bandwidth" means more hardware horsepower to compress and decompress the voice on the call. Race "The Tyrant" Van der Decken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg Sent: 16 December 2004 12:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Calculating required bandwidth I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Rather than asking for a full education here, can somebody point me at a suitable practical reference? Of course, if somebody wants to actually post the answer that'd be fine too :) THanks, ___ 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 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
