Discussion:
SIP Trunking Gateway
Tommy Laino
2012-07-03 16:20:11 UTC
Permalink
I am recommending a SipX to a customer. They are going to be
converting from PRI to SIP trunking. They want to have 24
trunks available for calls. I am assuming that the internal
SipX bridge is not going to be sufficient for the amount of
calls that they are looking to handle. I am looking for
recommendations from the forum for the best way to handle
all 24 SIP trunks including best hardware and SIP providers
for this deployment(I have used VoIP.ms and DIDLogic in the
past). Any help is greatly appreciated.
--
Tommy Laino
Dome Technologies
Matt White
2012-07-03 16:36:34 UTC
Permalink
Sipxbridge is more than capable of handling 24 simultaneous calls through a siptrunk if the hardware has decent specs ie...at least dual core and 8GB of ram.

We use Appia for all deployments. The load you noted here will be fine. they offer trunks with or without T1's.

Your biggest question is what to use for your internet connection. If they will have 24 simultaneous calls your looking at about 2048 KB if using G.711U (86KB with payload plus overhead). Which means you will need to move to G.729 to fit your max call bandwidth inside a T1.

-M
I am recommending a SipX to a customer. They are going to be
converting from PRI to SIP trunking. They want to have 24
trunks available for calls. I am assuming that the internal
SipX bridge is not going to be sufficient for the amount of
calls that they are looking to handle. I am looking for
recommendations from the forum for the best way to handle
all 24 SIP trunks including best hardware and SIP providers
for this deployment(I have used VoIP.ms and DIDLogic in the
past). Any help is greatly appreciated.
--
Tommy Laino
Dome Technologies
Tony Graziano
2012-07-03 16:56:50 UTC
Permalink
I'll ditto the price preference on that call volume. We would prefer not to
deal with trunk or internet quality if we don't have to.
Post by Matt White
Sipxbridge is more than capable of handling 24 simultaneous calls through
a siptrunk if the hardware has decent specs ie...at least dual core and 8GB
of ram.
We use Appia for all deployments. The load you noted here will be fine.
they offer trunks with or without T1's.
Your biggest question is what to use for your internet connection. If
they will have 24 simultaneous calls your looking at about 2048 KB if using
G.711U (86KB with payload plus overhead). Which means you will need to
move to G.729 to fit your max call bandwidth inside a T1.
-M
I am recommending a SipX to a customer. They are going to be
converting from PRI to SIP trunking. They want to have 24
trunks available for calls. I am assuming that the internal
SipX bridge is not going to be sufficient for the amount of
calls that they are looking to handle. I am looking for
recommendations from the forum for the best way to handle
all 24 SIP trunks including best hardware and SIP providers
for this deployment(I have used VoIP.ms and DIDLogic in the
past). Any help is greatly appreciated.
--
Tommy Laino
Dome Technologies
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: ***@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
Nathaniel Watkins
2012-07-03 17:36:09 UTC
Permalink
+1 - if reliability/quality are important (especially if you're looking for 24 (well, 23) channels) - PRI is the way to go. We have some call limit features in place on our Patton PRI gateways - if we have over xx channels in use, we can force outbound calls to be sent via a sip trunk (registered on the PRI gateway) to keep a few channels dedicated to inbound calling (this should only be an issue if you're worried about ever having more than 23 calls simultaneous calls the PRI) - basically, the sip trunks becomes your overflow for any outbound calls.


From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Tuesday, July 03, 2012 12:57 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] SIP Trunking Gateway


I'll ditto the price preference on that call volume. We would prefer not to deal with trunk or internet quality if we don't have to.
On Jul 3, 2012 12:27 PM, "Matt White" <***@thesummit-grp.com<mailto:***@thesummit-grp.com>> wrote:
Sipxbridge is more than capable of handling 24 simultaneous calls through a siptrunk if the hardware has decent specs ie...at least dual core and 8GB of ram.

We use Appia for all deployments. The load you noted here will be fine. they offer trunks with or without T1's.

Your biggest question is what to use for your internet connection. If they will have 24 simultaneous calls your looking at about 2048 KB if using G.711U (86KB with payload plus overhead). Which means you will need to move to G.729 to fit your max call bandwidth inside a T1.

-M
I am recommending a SipX to a customer. They are going to be
converting from PRI to SIP trunking. They want to have 24
trunks available for calls. I am assuming that the internal
SipX bridge is not going to be sufficient for the amount of
calls that they are looking to handle. I am looking for
recommendations from the forum for the best way to handle
all 24 SIP trunks including best hardware and SIP providers
for this deployment(I have used VoIP.ms and DIDLogic in the
past). Any help is greatly appreciated.
--
Tommy Laino
Dome Technologies
_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/

_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

________________________________
This message and any files transmitted with it are intended only for the individual(s) or entity named. If you are not the intended individual(s) or entity named you are hereby notified that any disclosure, copying, distribution or reliance upon its contents is strictly prohibited. If you have received this in error, please notify the sender, delete the original, and destroy all copies. Email transmissions cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Garrett County Government therefore does not accept any liability for any errors or omissions in the contents of this message, which arise as a result of email transmission.


Garrett County Government,
203 South Fourth Street, Courthouse, Oakland, Maryland 21550 www.garrettcounty.org
m***@grounded.net
2012-07-03 18:01:20 UTC
Permalink
Post by Matt White
Your biggest question is what to use for your internet connection. If they
will have 24 simultaneous calls your looking at about 2048 KB if using
G.711U (86KB with payload plus overhead). Which means you will need to
move to G.729 to fit your max call bandwidth inside a T1.
Appia told me all of their trunks are G.729 if I remember correctly. By the way, does anyone know what the total amount of bandwidth would be per G.729 channel with overhead?
Post by Matt White
-M
I am recommending a SipX to a customer. They are going to be
converting from PRI to SIP trunking. They want to have 24
trunks available for calls. I am assuming that the internal
SipX bridge is not going to be sufficient for the amount of
calls that they are looking to handle. I am looking for
recommendations from the forum for the best way to handle
all 24 SIP trunks including best hardware and SIP providers
for this deployment(I have used VoIP.ms and DIDLogic in the
past). Any help is greatly appreciated.
Matt White
2012-07-03 18:33:00 UTC
Permalink
Post by Matt White
Your biggest question is what to use for your internet connection. If they
will have 24 simultaneous calls your looking at about 2048 KB if using
G.711U (86KB with payload plus overhead). Which means you will need to
move to G.729 to fit your max call bandwidth inside a T1.
Appia told me all of their trunks are G.729 if I remember correctly. By the way, does anyone know what the total amount of bandwidth would be per G.729 channel >with overhead?
Appia doesn't hard code the codecs at all (if they limited it to G.729, AA/VM wouldnt work...its ulaw only). Rather its negotiated via the SDP. They allow both g729 and G711u. So you can choose your codec based on what you allow via sipxbridge and the codec order in the endpoint.

G.729 is about 32k with packet overhead.

Here is the reference i always use for codecs: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml
m***@grounded.net
2012-07-11 13:28:27 UTC
Permalink
Post by Matt White
Appia doesn't hard code the codecs at all (if they limited it to G.729,
AA/VM wouldnt work...its ulaw only). Rather its negotiated via the SDP.
They allow both g729 and G711u. So you can choose your codec based on what
you allow via sipxbridge and the codec order in the endpoint.
It's sometimes hard to get a proper answer from those guys.
Post by Matt White
G.729 is about 32k with packet overhead.
Their sales person insists adamantly that this is in fact 24K/call. Won't hear any other number.
Tony Graziano
2012-07-11 14:09:22 UTC
Permalink
They don't think Ethernet overhead. After you wrap the packet its 32k. Why
would you care what their sales monkey says?
Post by m***@grounded.net
Post by Matt White
Appia doesn't hard code the codecs at all (if they limited it to G.729,
AA/VM wouldnt work...its ulaw only). Rather its negotiated via the SDP.
They allow both g729 and G711u. So you can choose your codec based on
what
Post by Matt White
you allow via sipxbridge and the codec order in the endpoint.
It's sometimes hard to get a proper answer from those guys.
Post by Matt White
G.729 is about 32k with packet overhead.
Their sales person insists adamantly that this is in fact 24K/call. Won't
hear any other number.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: ***@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
m***@grounded.net
2012-07-11 16:26:49 UTC
Permalink
Post by Tony Graziano
They don't think Ethernet overhead. After you wrap the packet its 32k. Why
would you care what their sales monkey says?
I don't and I'm tired of his rude and know it all attitude so I've given up on them after wasting months of my time.
I am closing my account with them for the few DID's we already moved there and search continues for another good provider.
Tony Graziano
2012-07-11 19:16:28 UTC
Permalink
and they just announced g722 support...
Post by m***@grounded.net
Post by Tony Graziano
They don't think Ethernet overhead. After you wrap the packet its 32k. Why
would you care what their sales monkey says?
I don't and I'm tired of his rude and know it all attitude so I've given up on them after wasting months of my time.
I am closing my account with them for the few DID's we already moved there and search continues for another good provider.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: ***@voice.myitdepartment.net
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: ***@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
Todd Hodgen
2012-07-03 16:46:25 UTC
Permalink
Personally, I like to continue to use PRI for customers when I can. Many
PRI quotes today compete well with SIP trunks. And, many PRI's today are
delivered with SIP between the IAD and the carrier. However, they manage
the voice quality, and it is no longer an issue for you to have to deal
with. Big Bonus from my perspective!

Patton, Audiocodes, Epygi, etc. all have Gateways that will convert that PRI
to SIP reliably.

As Matt states, the sipxbridge can handle that number of calls fine.

You can provide some diverse routes with some SIP trunks as well. VOIP.ms,
since it is a prepaid service serves nicely for International calling, and
provides loss mitigation in case of being hacked. It also works well for
your conference bridge for when you need a bunch of extra trunks.

-----Original Message-----
From: sipx-users-***@list.sipfoundry.org
[mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Tuesday, July 03, 2012 9:20 AM
To: sipx-***@list.sipfoundry.org
Subject: [sipx-users] SIP Trunking Gateway



I am recommending a SipX to a customer. They are going to be converting from
PRI to SIP trunking. They want to have 24 trunks available for calls. I am
assuming that the internal SipX bridge is not going to be sufficient for the
amount of calls that they are looking to handle. I am looking for
recommendations from the forum for the best way to handle all 24 SIP trunks
including best hardware and SIP providers for this deployment(I have used
VoIP.ms and DIDLogic in the past). Any help is greatly appreciated.
--
Tommy Laino
Dome Technologies
Loading...