[cisco-voip] Authenticating sip trunk to ITSP from CUBE?
Dave Goodwin
dave.goodwin at december.net
Wed May 4 17:53:43 EDT 2016
Is there anything wrong with adding voice-class sip bind commands to ALL
the voip dial-peers, and then set the global binding to the interface that
faces the ITSP requiring authentication (since it seems sip-ua REGISTER
messages use the global bind)?
-Dave
On Wed, May 4, 2016 at 4:22 PM, Nick Barnett <nicksbarnett at gmail.com> wrote:
> Thanks for everybody's ideas.
>
> Unfortunately, 15.6 is OUT because it is not on the CVP 10.0 compatibility
> matrix :(
>
> I'm going to look at using multiple registrars and see if I can trick it
> into behaving... if that doesn't work, I guess I'll have to remove my
> global binding...
>
>
> On Wed, May 4, 2016 at 11:35 AM, Sreekanth <sreenara at cisco.com> wrote:
>
>> Yes, sip-ua tells CUBE to send REGISTER messages towards a Registrar
>> server globally with the authentication and credential parameters. These
>> REGISTER messages will be bound to the interface that is bound under voice
>> service voip -> sip. However, in the 15.6(2)T version, the tenant
>> configurations under the dial-peers will instruct the CUBE to send out
>> REGISTER messages.
>>
>> I just checked with the router in my lab and actually, option 2 won't be
>> possible. It won't instruct the CUBE to send out REGISTER messages. It will
>> only instruct the CUBE to add authentication credentials and realm settings
>> when sending out the INVITE messages towards the session target configured
>> under the dial-peer.
>>
>> You will have to go with option 1.
>> *1. Create the voice class tenant for the SIP trunk to ITSP and bind it
>> with the right interface.*
>> voice class tenant 1
>> registrar dns:cisco.com expires 3600
>> credentials username cisco password cisco realm cisco.com
>> authentication username cisco123 password 7 cisco123
>> sip-server dns:cisco.com
>> bind control source-interface GigabitEthernet0/2
>> bind media source-interface GigabitEthernet0/2
>> early-offer forced
>>
>> *2. Apply the voice class tenant to the dial-peer. Create specific
>> dial-peers towards ITSP.*
>> dial-peer voice X voip
>> voice-class sip tenant 1
>>
>> When this is done, CUBE will send REGISTER messages as well towards this
>> ITSP with the traffic bound to gig0/2.
>> This way you can have multiple ITSP trunks on 1 CUBE.
>>
>> Sreekanth
>>
>>
>>
>>
>> On Wednesday 04 May 2016 09:29 PM, Nick Barnett wrote:
>>
>> I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally
>> upgrade. Was actually planning on going to 15.4 this weekend. Jumping 3
>> versions kind of scares me, so maybe staging is in order.
>>
>> *I do have some limited auth commands on the dial peer, if this is what
>> you were talking about... but I don't think it applies in this scenario. I
>> don't have any options for credentials:*
>> CUBE(config-dial-peer)#voice-class sip authenticate ?
>> redirecting-number Use redirecting number credentials while
>> authenticating
>>
>> CUBE(config-dial-peer)#voice-class sip cred
>> CUBE(config-dial-peer)#voice-class sip c?
>> call-route calltype-video conn-reuse copy-list
>>
>> *There is also the registration commands:*
>> CUBE(config-dial-peer)#voice-class sip registration ?
>> passthrough SIP Registration Passthrough Options
>>
>> CUBE(config-dial-peer)#voice-class sip registration passthrough ?
>> dynamic SIP Registration Use dynamic Registrar Details
>> (default)
>> local-fallback Local Fallback - (e2e)
>> rate-limit SIP Registration pass-through rate-limit Options
>> reg-sync Registration Sync - send REGISTER when registrar up
>> (p2p)
>> registrar-index Registrar Index(s) used for registration passthrough
>> static SIP Registration Use static Registrar Details
>> system Use global registration passthrough CLI setting
>> <cr>
>>
>> *I tried using the system passthrough setting, but it did not work.*
>>
>> *I need to make sure I understand what is actually happening.*
>>
>> *I don't think the CUBE is even looking at dial-peers for REGISTER
>> messages. Am I correct? If so, no amount of dial peer settings is going to
>> make any difference here... unless there is a way to create a dial-peer
>> that will intercept REGISTER messages. I believe it is using the REALM
>> settings in the credentials and authentication strings (that I entered into
>> sip-ua). And sip-ua is using the global bind settings I set within voice
>> service voip -> SIP (which are set to the inside interface).*
>>
>> *Please set me straight!*
>>
>> Thanks,
>> Nick
>>
>> On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan (sreenara) <
>> <sreenara at cisco.com>sreenara at cisco.com> wrote:
>>
>>> What IOS version are you running on the CUBE? I can think of a couple of
>>> things.
>>> 1. In 15.6(2)T, a new feature has been introduced called multi-tenant
>>> where you can configure separate voice class tenants. Each tenant can have
>>> separate authentication mutually exclusive to one another and can be bound
>>> to different interfaces.
>>>
>>> 2. In your current IOS, check if you are able to configure the
>>> authentication and credential commands at the dial peer level. I am not
>>> sure which IOS had this introduced but it is worth a try.
>>>
>>>
>>>
>>> Sreekanth
>>>
>>> Sent from a phone.
>>>
>>>
>>> -------- Original message --------
>>> From: Nick Barnett <nicksbarnett at gmail.com>
>>> Date: 5/4/16 8:03 PM (GMT+05:30)
>>> To: Brian Meade <bmeade90 at vt.edu>
>>> Cc: Cisco VoIP Group <cisco-voip at puck.nether.net>
>>> Subject: Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?
>>>
>>> I'm binding control and media to my inside interface:
>>>
>>> sip
>>>
>>> bind control source-interface GigabitEthernet0/0
>>> bind media source-interface GigabitEthernet0/0
>>>
>>> I suspect this is the issue... is there any way to make the REGISTER
>>> messages come from the outside gi0/1 interface?
>>>
>>> The reason I'm binding to inside is that we have a a very fluid internal
>>> network. I have to make and modify internal dial peers almost daily. When
>>> I need to create a dial peer and put the bind statements on the dial peer,
>>> it won't bind properly since there are active SIP calls on the CUBE... so I
>>> bound it globally. My external dial peers rarely change, so I bind those
>>> directly to gi0/1 (on the DP).
>>>
>>> I was under the impression that REGISTER events can take place without a
>>> dial peer... but is there a way to, i dunno, make a dial peer for register
>>> messages? Can I use SIP profile magic to get it working as is?
>>>
>>> I found this article which is pretty much exactly what I'm dealing with,
>>> but it doesn't mention REGISTER at all...
>>>
>>>
>>> <https://urldefense.proofpoint.com/v2/url?u=https-3A__supportforums.cisco.com_blog_154506&d=CwMFAg&c=M-KQspD_LQogCbR-BWCHOaeDEPOhF8vWqHZTaiwxT3c&r=T9uVLZucbHG2NKKKzOrp-o5cpdReHj02PkJJsCVkgfwcv7S0R5lDeFJg2VRbiNih&m=UIAzGDQs8RCZld9kCbExwqpJhTgzpDVwM0k8_I7JRqU&s=jZN-R2pRsZOWN3r5is-aSivDlf9hqddUzDIoOWRWc3E&e=>
>>> https://supportforums.cisco.com/blog/154506
>>>
>>>
>>>
>>> On Wed, May 4, 2016 at 9:06 AM, Brian Meade <bmeade90 at vt.edu> wrote:
>>>
>>>> Do you already have the SIP bind under voice service voip?
>>>> voice service voice
>>>> sip
>>>> bind all source-interface FastEthernet0
>>>>
>>>> On Wed, May 4, 2016 at 9:58 AM, Nick Barnett < <nicksbarnett at gmail.com>
>>>> nicksbarnett at gmail.com> wrote:
>>>>
>>>>> I've never dealt with an authenticated SIP trunk before and I'm having
>>>>> some issues. I was wondering if anyone has had a similar experience. I
>>>>> already have 2 SIP trunks from ITSP-1 that do NOT require authentication.
>>>>> These are working fine and have been for years.
>>>>>
>>>>> We are adding ITSP-2 and their SIP service DOES require auth. I've
>>>>> followed their integration guide (which left a lot to be desired) and their
>>>>> acceptance team is telling me my auth is coming from our private class A
>>>>> address.
>>>>>
>>>>> Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP
>>>>> address. They are seeing REGISTER messages sourcing the inside VIP.
>>>>>
>>>>> I was looking around for an auth BIND statement or something like
>>>>> that, but I haven't had any luck. Any pointers?
>>>>>
>>>>> Thanks,
>>>>> Nick
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>
>>>
>>
>>
>
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