[Asterisk-Users] Calculating required bandwidth

[Damon Estep]

"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: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] 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 version 			of 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: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] 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,
> </edg>
> 
> 
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