[cisco-voip] Verizon SIP vs. PRI?
Nick Matthews
matthnick at gmail.com
Tue Feb 8 20:15:52 EST 2011
SIP trunks can be a blessing or a burden. Things that become important with
SIP trunking:
-All of your versions and VOIP applications become points of
interoperability. With PRI's, your interop stopped at the PRI. This means
contact center applications, voicemail, call control, etc. This even
extends out into the provider cloud - sometimes you'll have interop issues
with only certain DIDs or companies on the other side of their network. SIP
is open and flexible, which is good and bad. Sometimes the fixes to these
problems are complex and require for people like the SIP provider to take
action, which you can't control. You may find your contact center software,
call control software, border element software, and provider have differing
levels of interoperability flexibility.
-Sometimes it is a wash. If you're worried about survivability at the
branch, you can take the money you may save by centralizing the PRIs and use
it to get another circuit and router at the site. Now you've got higher
branch survivability and more bandwidth as well. And if you lose two
routers and/or two circuits at the site - you've probably got bigger
problems. The easy generalization is that you inherit the flexibility of IP
networks and get to work with the equipment you've invested into the network
rather than 20+ year old telephony technology. Since you've got increased
flexibility, it may be worth it to just flip the calls from the failed site
somewhere else until they've recovered.
-Faxing/911/Modems. Now that your VOIP domain is extended to the SIP cloud,
you have to take care to make sure you're standards based and compliant with
the provider. It's common for people to leave a small percentage of TDM at
the branch sites for 911, faxing, and survivability. Expect trouble here.
-Many choose to co-reside their SIP provider with their MPLS provider. This
prevents having a 'dumb pipe' for your SIP traffic with the capability to
distort the traffic without a disincentive. Imagine calling your cable
modem provider at home and telling them "you're causing 20% jitter and a 1%
loss on my high priority EF traffic". That being said - the internet is
quite significantly more reliable than many expect for carrying SIP
traffic. Skype, google voice, and a number of others are prime examples.
That being said - it's really cool. There are a couple places where it's
just awesome. If you've got a remote branch and the only voice offering is
a dusty T1 CAS circuit? Forget about that. If you have highly seasonal or
even unpredictably bursty traffic, it can be great. If you have a lot of
offices where you've overprovisioned the phone lines, SIP is a big cost
saver. It's portable, and you are no longer tied down to a single smart
jack where your T1 comes in on. When I travel, I register a SIP agent on my
cell phone to a HTTP PBX, which registers to a SIP trunk. It's the exact
same concept and architecture that SIP trunking for enterprises build on,
but cool-ified.
If SIP is confusing, just think about how an H.323 gateway works. Your CUCM
points to an IP address in your network. Now move your gateway out onto the
internet. H.323 and SIP are incredibly similar, so just switch the protocol
from H.323 to SIP. The last step is to place a Session Border Controller
(SBC) or in Cisco's terms a Cisco Unified Border Element (CUBE). This is
basically a voice firewall, and makes sure your signaling and voice stays
secure and is manageable. What really happens is that is takes the call,
terminates it, and then re-originiates it going outbound. From your CUCM's
perspective - it wouldn't know whether a PRI or another SIP leg was on the
other side.
In my opinion, once you get the hang of how SIP works, the troubleshooting
can be simpler too. Call quality problems and resolution can be a pain on
TDM circuits - who is to say where the distortion is coming from. With IP,
you can easily prove where the distortion is coming from with a sniffer.
Ever have to troubleshoot ISDN Q.921 messages? No thank you. It's a
stronger protocol than H.323 - the odd TCP handshake isn't a problem, it
isn't inhibited by a large specification, and it's a whole ton easier to
read and troubleshoot.
Signaling interoperability problems can be solved by going through the RFCs,
which can be muddy and fruitless - Vendor A allows only strictly formed
messages and Vendor B refuses to implement such details. Vendor A knowingly
doesn't follow SIP RFC #17, while Vendor B expects compliance. Vendor A is
written by a guy in Russia who quit 3 months ago, and Vendor B doesn't like
the way it was written, but Vendor B thinks it's correct. This was all
solved by TDM PRI's which are essentially the lowest common denominator of
voice termination.
Hopefully this gives you an idea. I suppose I'm just full of opinion on
this one.
-nick
On Tue, Feb 8, 2011 at 7:08 PM, Paul <asobihoudai at yahoo.com> wrote:
> Are you using CCM session management edition for that customer or is it
> just one cluster with a lot of centralized SIP trunks? I'm just asking out
> of curiosity since I've never heard of anyone using session management
> edition....yet.
>
>
>
> ------------------------------
> *From:* Bob Zanett (US) <bob.zanett at us.didata.com>
> *To:* Matthew Loraditch <MLoraditch at heliontechnologies.com>; David Eco <
> david.eco at msn.com>; "matthew at ciscovoiceguru.com" <
> matthew at ciscovoiceguru.com>; "notariannil1 at scranton.edu" <
> notariannil1 at scranton.edu>
> *Cc:* "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
> *Sent:* Tue, February 8, 2011 1:08:13 PM
>
> *Subject:* Re: [cisco-voip] Verizon SIP vs. PRI?
>
> I have helped to design, architect and deploy multiple customers with
> SIP, including SIP trunks carrying contact center calls and also business
> user calls. I helped to build out the initial infrastructure of a global
> medical device firm that almost has all of their US remote offices –
> literally 100s – migrated onto their centralized SIP trunks.
>
>
>
> Are there interoperability issues? Yes, as each vendor can speak a
> slightly different dialect. However, with the onset of some very nice
> Border Gateways – it is making this a non-issue. However, as Matthew states
> below – you need to work with someone who knows what they are doing.
>
>
>
> For instance, ROI can be a grey area. Many times, it is simply a wash and
> at other times can save a tremendous amount. Like the example above, each
> of the remote sites had a least a PRI. However, many of the offices hardly
> used 12 b-channels. By centralizing, they are seeing a very nice savings
> and they get:
>
> · Centralized dial plan management
>
> · Centralized DID control and management
>
> · On-the-fly call volume adjustment – it is now bandwidth not
> copper in the ground
>
> · Economy of scale
>
>
>
>
> http://www.callcentertimes.com/LatestNews/tabid/59/ctl/NewsArticle/mid/407/CategoryID/1/NewsID/173/Default.aspx
>
>
>
> Cheers,
>
>
>
> Bob
>
>
>
> Solutions Architect
>
> Customer Interactive Solutions
>
> Dimension Data
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Matthew
> Loraditch
> *Sent:* Tuesday, February 08, 2011 2:29 PM
> *To:* David Eco; matthew at ciscovoiceguru.com; notariannil1 at scranton.edu
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Verizon SIP vs. PRI?
>
>
>
> The thing I would say is it all depends on your level of comfort with the
> Carrier and with SIP. I generally don’t have a problem with the quality of
> the service when it works, the problem has been troubleshooting and making
> it work in the first place. I understand how to debug and troubleshoot a T-1
> or PRI and can set one up H.323 or MGCP no problem. SIP just doesn’t process
> for me, Everybody says it’s simple, plain text and all that but I don’t get
> it.
>
>
>
> My point, If you are going with it make sure you or someone you have
> understands it.
>
>
>
> If you have a smaller install one thing you can do is have the provider put
> in an IAD and hand off as PRI. Make them responsible and make your side
> something you understand. Obviously in a large campus environment that may
> not be feasible and defeats some of the consolidation benefits but in a
> smaller setup it can make life much easier.
>
>
>
> *Matthew Loraditch, CCVP, CCNA, CCDA*
> 1965 Greenspring Drive
>
> Timonium, MD 21093
> support at heliontechnologies.com
> (p) (410) 252-8830
> (F) (443) 541-1593
>
> Visit us at www.heliontechnologies.com
> Support Issue? Email support at heliontechnologies.com for fast assistance!
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *David Eco
> *Sent:* Tuesday, February 08, 2011 2:47 PM
> *To:* matthew at ciscovoiceguru.com; notariannil1 at scranton.edu
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Verizon SIP vs. PRI?
>
>
>
> with SIP, you may get more capacity of the channel. But in our case,
> we have to setup IPSEC VPN with Verizon to utilize their SIP Trunk. Network
> has to be reliable.
>
> David
>
>
>
>
> > From: matthew at ciscovoiceguru.com
> > Date: Tue, 8 Feb 2011 13:43:34 -0600
> > To: notariannil1 at scranton.edu
> > CC: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Verizon SIP vs. PRI?
> >
> > One of my biggest questions right now regarding SIP trunks has been data
> circuit issues and usability in SRST mode. If your branch office loses it's
> network connection, will your SIP trunk go down too? If so, then you're up
> the creek without a paddle.
> >
> > Sent from my iPhone
> >
> > On Feb 8, 2011, at 1:13 PM, Lisa Notarianni <notariannil1 at scranton.edu>
> wrote:
> >
> > > We currently utilize PRI trunks as fail over and backup. They connect
> to our 6509's.
> > >
> > > Can anyone share their thoughts on SIP vs. PRI services.
> > >
> > > Thanks,
> > >
> > > Lisa
> > > --
> > > <LNsignature.jpg>
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ------------------------------
>
> [image: DDIPT] <http://dimensiondata.stream57.com/04141pm/>
>
> * Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> ------------------------------
> Don't get soaked. Take a quick peak at the forecast
> <http://tools.search.yahoo.com/shortcuts/?fr=oni_on_mail&#news>
> with theYahoo! Search weather shortcut.<http://tools.search.yahoo.com/shortcuts/?fr=oni_on_mail&#news>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110208/2963d58e/attachment.html>
More information about the cisco-voip
mailing list