I was part of the team that starting a large scale sip migration
almost 10 years ago. Have moved thousand's of DID since then. Run
multiple gig circuits into the cube.
Recommendations:
on the link to your provider, use address outside of the route able
block for your company. (say you use 10.x.x.x then use 172.16 or
192.168) If you can, don't route the itsp connections on your
company network, go direct to the routers supporting those links.
(BGP peers I would guess depending on carrier/build) If you can use
a dedicated router, unless is a small site.... This is important if
you wind up doing any kind of call recording, or if you have to enable
debugs during the day.
Use dedicated dial peers setup exactly for each itsp SBC link for in
and one for out.
Use something like the "voice class uri trunk(x) sip" or equivalent
to bind to the dial peers for each SBC.
This will help if you have to add additional carriers, or say
acquire a company, or need to do special routing...
use full E164 to and from the carrier, they may only want to do 10
digit in/out, but that is easy enough. (uri trunkx will help here, as
the inbound number will be at the cube, then you can route to cucm
with outbound dial peer)
From your CUCM still send the 9 or 8 or whatever for outbound, then
strip on match in the dialpeer to Itsp. This will keep call looping
etc.
define your voice class codecs on the dialpeers... don't just assume
it will take the default, or work as you want without it.
if the cube will never see VIDEO, disable the options. The cube
software likes to release bugs that cause the cube to go south with
video errors.
Depending on your carrier, you may need to force G729 or G711 first,
even if its not your preferred codec, have seen were the SBC will not
negotiate a call, if the codecs aren't in the order the carriers SBC
wants.
do not assume the carriers network will normalize the calls. For
instance, if the destination is on the same carrier, its a direct ip
route via the SBC. If that end side can't accept say G729 (cheaper
sbc) the call will just fail.
NEVER user debug ccsip all
debug CCSIP messages is safer, and unless your cube is peeked, it
won't add to much cpu.
make sure your CPU never exceeds 80% at the max possible peek of routing.
Check how the calls work with MOH. Inbound and out. make sure 2 way
audio remains after the on hold event..
Do you need to force early offer? (yes sounds silly, but have run into
issues where some phones had no audio unless this was set)
Ask your carrier, how they handle TFNs outbound, if you pass the ANI
from a 3rd party. (this is all billing stuff to the TFN owner)
Some may allow calls to process not caring what the number is.
Some may allow you to provide a alternate billing number.
Some will just 603 decline the call if the ANI isn't in your
number poll assigned to you.
with a 603 the cube will try the next dial peer so you can add
a header to re-write this with your number.....
Diversion headers exist, however most carriers pass them through to
the destination, and IVRs or Voice Mail systems on the far side will
try to process that information, and do unexpected things. (the party
your calling doesn't exist for example.)
define the default sip control/media source interface, this will be
your destination from cucm. The URI trucks will define the sip
control/media on the ITSP side.
If you use firewalls any where in your company, that will pass
voip... Set the rtp-port range on the cube match the smaller range
of what your going to use. (say the old days 16384-32767) don't assume
the firewall will pass all the UDP ports by default.
speaking of firewalls, check, double check, and triple check, then do
your own research if you will use them, when it comes to SIP
inspection. Some FW's have options that need to be tweeked and
defined, for the SIP port. (this may control anything from timeouts,
which media ports engage) This is especially true with expressway
in the DMZ. It might be safer to not use sip inspection and just pass
the port. But for some FWs this is not true.
define the FAX-relay, rats and protocols for T38
ask your carrier how they handle QOS. some don't since the trunk to
them might be dedicated.
use option pings on the dial peers, so if the SBC goes away that
dialpeer disables. The sbc side just has to respond, even if its an
error saying what is this... that will keep the peer up.
Setup the event manager applet. have it email you on syslog patterns
for dialpeer status. Then you will know if the link goes down.
if you can get a bug scrub on the version of IOS, don't be determined
to use the newest code. newest is not always best.
Hope at least one thing here was helpful.
On 2/10/22 9:09 AM, Matthew Huff wrote:
We are in the process of migrating for a legacy PTSN voice gateway
(PRI) to a new CUBE based SIP connection to a iTSP connected via a
private metro ethernet (not Internet based). Does anyone have a good
source for recipes / dial-plans recommendations / best practices for
this?
*Matthew Huff*| Director of Technical Operations | OTA Management LLC
/Office: 914-460-4039/
/mh...@ox.com <mailto:mh...@ox.com> | //www.ox.com <http://www.ox.com>/
*...........................................................................................................................................*
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