SIP to DS3

[Oliver Dilan]

Hi I currently have a Avaya conferencing bridge that has a T3 interface card we use to run a voice DS3 to the card but want to do away with that and go SIP to DS3. I know some people who are using Cisco setup where they use this as the SIP gateway so there just running a Fiber link to the Router have a T3 module card on the Router since there not terminating a voice DS3 on the router I don't believe its channelized.  So basically there's sip carrier sends the call to the router the router then passes a call through the DS3 module card down to the conferencing Bridge what I want to know is what equipment do I need to support a DS3 and do I need any special voice software. I have heard I don't need the voice software but it's a little bit out of my realm.

[Philip D'Ath]

Are you sure you can't get a SIP card for you Avaya conference system?  That would be so much easier ...

I don't know the exact answer, but you'll need a Cisco 4451 and a SM-X-1T3, and enough DSP resource for the number of channels that you want.

http://www.cisco.com/c/en/us/products/collateral/routers/4000-series-integrated-services-routers-isr/qa_c67-728547.html

I'd probably go for the Voice bundle, ISR4451-X-V/K9, which comes with a bunch of DSP resource.

http://www.cisco.com/c/en/us/products/collateral/routers/4000-series-integrated-services-routers-isr/guide-c07-732797.html

[Oliver Dilan]

This is a Older Avaya Spectel cs700 TDM based bridge comes with 10 DSP cards and 3 DS3 boards. On the router I will be using a 1-gig fiber bandwidth there's no voice DS3 going into the router. The way I saw it was my sip carrier sending the call to the router and then the router pushing the call down to the bridge I didn't know you needed DSP resources for that.

[Philip D'Ath]

My understanding is, you need a DSP resource for every channel you want to process.  If a channel comes in on SIP it needs to be processed and sent out the DS3.  If the codecs don't match then it would also need to be transcoded which could consume even more DSP resource.

[Oliver Dilan]

So let me get this right I have a 3945 Cisco router. That router has a fiber Uplink at 1gig. My carrier who handles my sip calls will send that call to my router my router than will process that call through sip and send it down to my bridge through the DS3 module card on my 3945 which I will run DS3 cable TX RX from Cisco to my conference bridge DS3  boards. Obviously a full DS3 - 672 channels. So to process all this I would need 2 pvdm3-256 1 pvdm3-128 and 1 pvdm3-32. Because I can get conferences at times with 600 concurrent calls

[Philip D'Ath]

The 4000 series routers replaced the 3900. If you had a 4451 a single SM-X-PVDM-500 can do 768 channels.

[Oliver Dilan]

Thanks for your help my last question I'm going to be buying three routers and the 4451 is the most expensive of the four series if I went with a 4331 or 4351 would not support the pvdm4 500 and the SM - x t3. I'm going to be doing one DS3 I need router so I figured I only need 53mbps 4 a full rate DS3.

[Philip D'Ath]

Alas the DS3 card is only supported in the 4451 (according to its data sheet).

There is also another DS3 card option as well.  Have a ready of the data sheet in the link I gave previously, and especially note the things it says it does not do - to make sure you wont have any issues.

[Oliver Dilan]

Phil I wanted to add this which I hot from another avaya engineer. 

All the carriers are sending the G711 uncompressed protocol to clients so there is no transcoding going on.  You can insist on only full speed protocols from the carrier.  The actual stream is around 80 kbps per call with overhead or worse case for a DS3 around 51.5 Mbps.  Once a SIP call is set up there is a pass through connection of the voice packets through the router they are not processing the voice stream only the call setup.  The Cisco 3845 has a maximum forwarding rate of 256 Mbps and a DS3 is over SIP is 51.5Mbps  That is why it is rated for two fully loaded DS3’s. The next model up the Cisco 3945 is rated at 500 Mbps which is overkill. Do you understand what there saying

[Philip D'Ath]

I haven't worked with voice for a while, and when I did it was with E1's.  Every SIP channel going through it needed DSP resource.  I don't see why T3 would be any different.

Can you get a sample config?  That is likely to reveal if it will need DSP resource.  If it is using dial-peers then I think it would need DSP resource.

I guess go with your Avaya guys.  If they have done it before, know the part numbers and the configuration then you need to trust their advice.

[Oliver Dilan]

Hi Phil I guess you were correct I see one pvdm2 you can correct me if I'm wrong but here's the inventory and config of what they have for a T1 can you tell how many dip resources they would have 

Oliver,

Here is the ‘show inventory’ of my 2851:

GC2851#show inventory

NAME: "2851 chassis", DESCR: "2851 chassis"

PID: CISCO2851         , VID: V01 , SN: FTX0935A1NT

NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"

PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC09260SBU

NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 1", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"

PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC092106FN

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSP SIMM with four DSPs"

PID: PVDM2-64          , VID: V01 , SN: FOC10350UCC

Here is the running-program of the 2851:

Current configuration : 6616 bytes

!

! No configuration change since last restart

version 15.1

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname GC2851

!

boot-start-marker

boot-end-marker

!

!

card type t1 0 0

card type t1 0 1

logging buffered 4096

no logging console

enable secret 5 $1$ttDq$0TdTWs1xF2YGwb.ZdLrj7.

!

aaa new-model

!

!

!

!

!

!

!

aaa session-id common

!

clock timezone Pacific -8 0

network-clock-participate wic 0

network-clock-participate wic 1

network-clock-select 1 T1 0/0/0

network-clock-select 3 T1 0/1/0

!

dot11 syslog

ip source-route

!

!

ip cef

!

!

!

ip domain name goconference.com

ip name-server 8.8.8.8

ip name-server 8.8.4.4

no ipv6 cef

!

multilink bundle-name authenticated

!

!

!

!

isdn switch-type primary-ni

!

!

trunk group Atlas1

hunt-scheme sequential both

!

!

voice call send-alert

voice rtp send-recv

!

voice service pots

!

voice service voip

ip address trusted list

  ipv4 74.119.8.70

  ipv4 74.119.8.86

  ipv4 74.119.8.120

  ipv4 199.87.44.10

  ipv4 74.119.9.138

  ipv4 204.10.92.138

  ipv4 204.10.92.137

  ipv4 74.119.9.137

!

voice class codec 711

codec preference 1 g711ulaw

codec preference 2 g729r8

!

!

!

!

!

voice-card 0

!

crypto pki token default removal timeout 0

!

!

!

!

license udi pid CISCO2851 sn FTX0935A1NT

username voip privilege 15 secret 5 $1$3hGe$kw3uB/hPSrmQJhwWUA7pU1

!

redundancy

!

!

controller T1 0/0/0

cablelength short 110

pri-group timeslots 1-24

description Internal_T1_1

!

controller T1 0/0/1

cablelength long 0db

!

controller T1 0/1/0

cablelength short 110

pri-group timeslots 1-24

description Internal_T1_2

!

controller T1 0/1/1

cablelength long 0db

!

!

class-map match-any CCP-Transactional-1

[Philip D'Ath]

A PVDM2-64 can do 64 x G.711 channels.

I don't see how that config would work - but it may be relying on some default behaviour.