Pros and cons of SIP vs PRI on IPO 500V2

[mikedphones]

We have a client who has persistent, issues with their SIP from Paetec who also provides their 3 MEG bonded T1. They have been great about troubleshooting with us, but it just is hit or miss sometimes. 2or 3 out of 10 calls. All the sets are digital so I am thinking this echo and drop as well as garbled voicemail messages from time to time are the call coming in. Once or twice is was real bad and they power cycled the IPO which seemed to clear it. My arguement was that since they bounced the connection, it cleared it and it isn't the IPO.

They have made firmware adjustments, we are running 729 compression now, still get the issue. They have QOS. We went from 6.08 to 6.14, going to 6.18 soon.

Paetec gave us own own handoff from the Adtran so we go direct to the LAN2 port on IPO and have their inside network on LAN1 to allow for admin and VM PRO connectivity. I can't imagine that the customer network has anything to do with it as we are direct.

I guess my question is if it's good a majority of the time, is the IPO and programming OK and it's just the nature of SIP???? Should we just go PRI? I have never had an issue with PRI on IPO.

I find PRI circuits to just work. I am considering having them change the handoff to PRI but wanted to give SIP a fair chance. It's been back and forth for 6 months now.

Any input appreciated.

 

[IPGuru]

Sip Pros:
Cheap calls

Sip Cons

PITA to configure (each sip provider seesm to be differnt)
call quality dependent on quality of customers data network
call quality dependent on customers internet connection
call quality dependednt on the rest of the internet.

PRI Pros:
well defined connection standards
reliable
exelent call quality

Pri Cons
Cost.




I do not Have A.D.D. im just easily, Hey look a Squirrel!

 

[kholladay]

SIP with Paetec, Qwest, Cbeyond etc is not the same as SIP from ITSPs like Flowroute, LES.net, etc. Paetec and the lot give you a dedicated data circuit for your voice (and data) with an Adtran or Cisco router that provides the split between Internet access data and the voice call traffic along with QoS. SIP voice traffic rides on the provider backbone directly to the SIP proxy/server and never hits the public Internet so, assuming you had a tech who knew what they were doing, voice quality should not be an issue. Also the setup for these types of circuts is extremely simple and in most cases you either plug the router directly into the WAN/LAN2 port of the IP Office or they request the IP address you've assigned to the IP Office and they route based on IP rather than port. In either case you have no issues unless you are going through a secondary router/firewall.

Voice qualtiy should not be an issue in either case. Well over 50% of our new installs are SIP through one of the providers above or one of several others who deliver a dedicated circuit and would rather hand off SIP direct vs dropping in an IAD and converting the native SIP circuit to a PRI (and yes if you order a circuit from Paetec or Cbeyond you'll get a SIP circuit 99% of the time and when you request PRI they simply conver the SIP to PRI using a piece of equipment on-site).

Most of the providers like to use G.729 by default but should negotiate to any other CODEC you request in the call INVITE if you have disabled G.729 as an available option. This is usually fine for voice calls but don't try to fax over it.

Sooooo. All that being said. SIP on the IP Office either works or doesn't work. If it doesn't work then you have configured it wrong or the provider has given you a tech who doesn't know what they are doing. If it does work then the occasional voice quality issues are usually a result of the router not doing what it should.

Have you checked with Paetec to get their input?



Kyle Holladay
ACSS SME Communications
ACE Implement: IP Office
MCP/MCTS Exchange 2007
Adtran ATSA, Aruba ACMA

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[IPGuru]

My 1st post looks like i was having a bit of a downer on sip

what I meant to imply is that when sip is installed on a quality circuit to a good supplier it works fine, but as always you get what you pay for
go for a cheap carrier with a cheap internet connection (basic ADSL?) & it is not going to be nice

I am not familiar with paetec so was not commenting on their provision (which sounds very good from kholladay's post)


I do not Have A.D.D. im just easily, Hey look a Squirrel!

 

[kholladay]

LOL @IPGuru. I wouldn't say Paetec is good or bad. They simply are, as all other traditional/legacy carriers are...well, they are who they are and let's leave it at that.

Different people have different experiences with SIP. I've been lucky to suffer through every change Avaya has made to SIP in IP Office since they introduced it which has made me familiar enough to comment in most situations.

Personally I like SIP and we switched from traditional service to SIP a few years back and have been happy.

Kyle Holladay
ACSS SME Communications
ACE Implement: IP Office
MCP/MCTS Exchange 2007
Adtran ATSA, Aruba ACMA

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[tlpeter]

SIP can be a pain in the ARS (sorry, a funny word joke) sometimes.
ISDN is far more reliable.
ISDN always works, no matter what you do (When the line is good)
Most SIP providers are different.
SIP needs to get more mature but then it should be better.


When you pay peanuts, you get monkeys!

I'm not insane, my mother had me tested!

 

[hairlessupportmonkey]

I think we should stop comparing SIP with ISDN, but compare how its delivered to the CP.

They are two different technologies that both have their pros and cons. we all know this.

The two biggest most crucial factors with delivering SIP based voice is 1) the provider and 2) the delivery method.

SIP doesn't need to get mature. It already is. The issue is being the delivery method is too mature. rotting copper in the ground trying to support high bandwidth for xDSL is the downfall.

lets put this into perspective. Ive seen ISDN delivered over old copper and its really bad. BT change it to fibre and its perfect. so if every one started using dedicated low contention fibre based BB then I think we will see SIP as reliable as ISDN

One other issue with SIP is people mistake it for a standard, its not, its a protocol and yes it might be nice if everyone played by some nice standard and in general many do, but its not standardised as such, but that's all dependent on the carriers back end, be it a Top tier provider with Genband M6's or a smaller type company reselling to you via their OpenSIP / Asterisk server in a data centre somewhere.

ACSS - SME

 

[mikedphones]

Thank you all for your input. Kyle, I guess I want to be most critical of myself and my programming before Paetec's. My first questions would come back to compression. They are now sending 711 instead of 729,. Either way, if I set the config to be AUTO SELECT is that safe? 711 is likely to be better if that's all they send. Bandwidth isn't an issue as they are moving off to FIOS or some other method for network and this 3 meg will remain just for SIP use and data redundancy.

I don't have the config in front of me but I do recall on the SIP Line Tab, for some reason Network Topology was NONE at one point rather than LAN2 which is where the Paetec hands off too. They got calls with NONE, but I would think it should be LAN2, Agree?

Also Layer 4 was UDP SEND 5060, When LAN2 is selected for Topology, 5060 grays out in the LISTEN PORT. Is that OK?

Thanks!

 

[tlpeter]

G.711 is better for call quality.
When it is set to None then you need an iproute.
If it is set to Lan2 you should use STUN.


BAZINGA!

I'm not insane, my mother had me tested!

 

[kholladay]

The LAN1/LAN2 are not necessarily where the traffic is landing but what topology information is presented in the outbound calls (basically how to tell the far end what our public IP is instead of the private IP so they can get back to us). If Paetec is connected directly to LAN2 you don't need to specify topology information as they know who/where you are. If you set it to none the system uses the IP route table to find its destination which is perfectly fine. Also if you are using registeration you usually don't need to provide topology information as the registration server knows the IP address that registered.

G711 provides much better voice quality so if bandwidth isn't an issue I'd go with that. Automatic is fine but I don't care much for it. I typically set mine to use G711 the G729 as the codecs in that order (you can deselect G723 and ALAW so they aren't used)

Also if you use automatic and Paetec is offering only G711 on the invite that is what you'll get. However on your outbound you'll offer the entire range of CODECs and they may accept a more lossy option unless you lock it down.

Kyle Holladay
ACSS SME Communications
ACE Implement: IP Office
MCP/MCTS Exchange 2007
Adtran ATSA, Aruba ACMA

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[jayse81]

We are about to install Paetecs Dynamic IP PORT T1 for SIP connectivity for a customer. What is the hand off from the circuit to the IP Office? It is an IP Office 4.2

 

[intrigrant]

In the near future it is silly to compare SIP with ISDN because the providers of ISDN are delivering the ISDN line over a IP connection, so you get a box on site converting SIP to ISDN.
What is then better? I think a SIP trunk because you do not need a extra box between the IP Office and the provider.

 

[Albus2]

To bring the discussion in another direction, what about features in SIP vs ISDN?
SIP simply does not have all features of ISDN. If the customer just want to answer incoming calls or place simple outgoing calls, then SIP is fine.

If he wants to divert to external destinations (while presenting the A number) or use Twinning (of course with presenting the A number on the mobile phone) then it starts to be difficult. If they need to present different numbers for outgoing calls like own DID, withheld, a group number, a privat number or even something special like a 0800 then you are off. At least when one user needs all or some of these options. Call cost is not available.
Not to forget some special devices like alarming systems using analog modems not working over SIP.

Saying that, it's clear that not every problem relies in SIP. Some are just because Avaya makes SIP configuration so different from ISDN. And changes it in every release.

I think of one of our first SIP only customers. All went well, we could find a way for everything.
But then there was a new user with a lot of special needs. No problem if we had ISDN. But with SIP... It's really a pain in the ARS.

 

[cwhitmore]

Mike,
Did you end up moving to PRI? We are having the same issue with Paetec, but don't have the spare bandwidth to test G.711.