[cisco-voip] SIP to iTSP best practices
Kent Roberts
dvxkid at gmail.com
Fri Feb 11 11:59:07 EST 2022
First off. if your cube is working and in production, don't just
disable video... you might be in for a surprise. It should be done in
a test window time where you can test phones/all software.
Jabber/webex supports video, are you using those? does your carrier
use/support it? Do they charge extra if you send them a video call?
(things to ponder)
This is a tricky one, and does, but does not have an easy answer....
First thing I would do is make sure the Retry Video Call as Audio check
box is (still) checked on your trunk.... (believe this is default...)
On the cube:
Cisco has a doc that calls out video suppression, two ways:
globally (IOS release dependent, later releases (15.62T and 16.3.1)
voice service voip
sip
audio forced
dial-peer level
voice-class sip audio forced
That was the easy side. You should be golden here....
But....
remember, your carrier may not like this..., so you might have to look
at sip-profiles instead. Use these to modify the data in/out on the
dial-peer to make your carrier happy, if you get 488 errors back. (or
others) you might need the carriers help to understand what the don't
like.... (There is all kinds of SBC's out there...) Most carriers do
have decent equipment monitoring there links, so you can always call
them and ask, why are you rejecting my call.... BUT keep in mind, the
carriers don't always have people trained in the area you need to help
with this, so asking them for a detailed SIP trace might be the best
idea, and call TAC for their thoughts on how to resolve.
sorry not exactly a straight answer, but if your not live yet, it will
be very easy to make happen..
On 2/11/22 8:52 AM, Matthew Huff wrote:
>
> Thanks for the recommendations. I have a lot to dig into. Question
> about the video disable. We have no video hardware, so think it would
> be good to disable it before we go live. What’s the best way to
> disable it globally?
>
> Is it
>
> Voice service voip
>
> Sip
>
> Audio forced
>
> ?
>
> *Matthew Huff*| Director of Technical Operations | OTA Management LLC
>
> /Office: 914-460-4039/
>
> /mhuff at ox.com | //www.ox.com <http://www.ox.com>/
>
> *...........................................................................................................................................*
>
> *From:* Kent Roberts <dvxkid at gmail.com>
> *Sent:* Thursday, February 10, 2022 6:14 PM
> *To:* Matthew Huff <mhuff at ox.com>; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] SIP to iTSP best practices
>
>
>
> 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/
>
> /mhuff at ox.com | //www.ox.com <http://www.ox.com>/
>
> *...........................................................................................................................................*
>
>
>
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>
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