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SIP trunking with a Cisco 3825 integrated services router

Posted on 2013-10-29
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Last Modified: 2013-11-11
We are trying to implement SIP trunking into our Interactive Intelligence system and I was hoping to use my Cisco 3825 integrated services router to perform this function. I read that it is possible.

I do not know what I am doing at all with SIP, but I do have my CCNA from 2002 (ok well it is expired).

My environment: Time Warner fiber @ 50m up/down burstable
Cisco DMZ switch (nothing special)

Client gateway: Untangle

1) What are the high level DMZ/router requirements as far as QoS?

2) What are my options on the IOS version and what do they afford me?

3) What physical modules does my 3825 need to have to perform all of this?


I am trying to do this on a relative budget, which is the whole reason we're trying to go SIP. Your wisdom would be appreciated. I am expected to start testing this circuit by next Friday. (It is October 29, 2013 today)
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Question by:Pauli311
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by:agonza07
ID: 39610741
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Author Comment

by:Pauli311
ID: 39612284
I can send links too, man. I'm asking for an experts opinion who understands the very elements I am asking.

I thought it was called Experts Exchange, not LINK exchange.


What I need to know is, how can we do this under the old IOS model before they switched to the current licensing scheme with our 3825?
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by:agonza07
agonza07 earned 500 total points
ID: 39612364
lmao, sorry, sounded like you just wanted to refresh your skills and needed a place to start :)

Even on the 2800/3800 series, you still need voice licensing. It's not like the new series where you need PAK to activate, so just having the IOS is all you need, but to be properly licensed you need that piece of paper telling you you're licensed.

If possible, try to have IOS v.15.1 and above, and go with IP Voice or Advance Enterprise.

For QoS, it depends on your SIP provider and how they will be delivering the trunks. Some like to put their own session border controller on site, and others do it over the internet. If they have their own equipment on site, then QoS should be taken care of by them, if you are going over the internet then there's really no QoS. You can prioritize the order of outgoing packets, but you can't control how they come in or reach the destination.

SIP is handled through ethernet ports, and unless you are doing PSTN conversions like terminating to a PRI or FXS ports, then you dont need anything special on the hardware side. If you are doing PRI or FXS/O then you'll need the appropiate cards and some PVDM modules.

Also, if you are doing codec translations or going to use you 3825 as a conference bridge, you'll need some PVDMs.

Let me know if you need further help.
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agonza07 earned 500 total points
ID: 39612374
BTW, here's another link. This is pretty much all there is to it on the configuration side plus a few dial peers.

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml

Pretty much just this:

voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
allow-connections h323 to h323

For incoming calls

dial-peer voice 1 voip
session target ipv4:10.13.8.150
incoming called-number 8...
dtmf-relay h245-alphanumeric
codec g711ulaw

For outgoing calls

dial-peer voice 2 voip
destination-pattern 8...
session protocol sipv2
session target ipv4:10.13.8.16
dtmf-relay rtp-nte
codec g711ulaw

Then if provider needs a username and password

sip-ua
registrar ipv4:10.1.1.10
or
registrar dns:csps.cisco.com
authentication username xyz password xyz realm cisco.com
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Author Closing Comment

by:Pauli311
ID: 39612379
This is the golden information I was looking for!

 Thank you for having a sense of humor, too.

Do you mind watching this thread in the next week? I don't expect lightning responses like you did today, but it is much appreciated! To be fair, I hate looking at any documentation Cisco puts out. It is like eating glue.  It is so terrible and hard on the eyes.
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Author Comment

by:Pauli311
ID: 39639909
I was able to upgrade to 12.4 (25), and we are going to test this week. Thanks again
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