CUBE and SIP question

[Bekzod Fakhriddinov]

Hi, we have one cube and sip to service provider(SP) , management is considering another redundant SIP trunk to be established via another l3mpls network we use for data traffic only . The thing is CUBE is not directly connected to that l3mpls network but connected via LAN . 

Is it possible to setup 2nd  SIP trunk to be established via another l3mpls network over LAN ? 

I don't know how to deal with IPs at this case, cause we are connecting to PUblic ip of the SP...

and also I don't have PVDM modules and CUBE license on the router connected to that l3mpls network...

Any idea, recommendations?  

Thank you 

[R0g22 · Accepted solution]

You would need all of those as well. How will the routerB route calls then ? So if basically can be the same as your existing CUBE. During failover, your existing CUBE will route calls to your routerB. This routerB will then route calls over the backup SIP trunk, so it needs to have a dial plan as well.

[Bekzod Fakhriddinov]

Nipun, great explanation, thank you , you are the best qualified, best patient engineer.  

[Bekzod Fakhriddinov]

May I ask you last question Nipun? I just checked routerB configuration , I thought it has the same config as another one, but this is connected to L3mpls Carrier  with our private IPs and BGP. So I will need to give sip carrier my private IP to connect . As I remember this is not recommended by Cisco, that's why I will exclude that ip/30 from ospf routing so that /30 subnet will not be routeable , but I still will be able to connect other L3mpls networks (on other sides) because they will be redistributed via BGP to OSPF (except that /30 network). 

This is kinda I am replacing public ip with dummy ip (non-routable ). 

What do u think ?

[R0g22]

Since you have L3MPLS, your routing would be done by your ISP either through per customer VRF or L3VPN. The private IP address that you will want to use in this case would be external or ISP facing. The NAT to a public IP would be done by a PE GW upstream. I would suggest you to talk to your MPLS ISP and see how things are setup. With SIP you need to take care of the embedded IP addresses within the L7 payload i.e. your INVITE etc. They need to be NAT'ed as well. If the PE GW does not support ALG, you will face issues.
Your ISP might ask you for your subnet in order to extend connectivity to your actual VRF.

To summarize, talk to your ISP. See how they are doing things and if you run SIPoMPLS what changes/additions need to be done at your or at their end.

[Bekzod Fakhriddinov]

Hi Nipun. I think I can use existing SIP trunk config on CUCM  to connect second SP (CUCM pointing to internal ip of CUBE). And also I see, I will need only add outboud dial-peers only to the new SIP on existing CUBE for this second SIP trunk/SP, because inbound Dial-peer is bind to interface and no session target ip of the SP , so it would accept sip packets from both sip providers.     We would like to keep both SIP trunks active . 

Correct? 

[R0g22]

Correct, you can use the same CUBE for both the trunks. If both the trunks need registration, you would need to use multi-tenant -
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-multi-tenants.html

You might need to upgrade the IOS on the router to make use of the feature.

Another thing, if both of your ITSP's agree to the same dial-plan, you don't need to create additional dial-peers. You can add a voice class server-group. You can then define the hunting within the server group to load balance etc.

[Bekzod Fakhriddinov]

Hi, If sip service provider gave me 2 of their IPs one for media and second for signaling I think I need to point outbound dial-peer session target to their signaling ip , correct?

Both IPs are from the same network so as soon as I establish sip signaling media will flow the same way. 

Also I guess "voice-class sip bind control source-interface GigabitEthernetx/x"  on inbound and outbound interfaces will help to properly route . 

Thank you 

[R0g22]

Important thing to note is that the bind command is relevant local to the router. So from the router's perspective, the media and control bind will be done for the interface that is ITSP facing/external. As far as the session target is concerned, it would point to the ITSP proxy server. They need to tell you that. It should be what they are referring to the signaling IP.

[Bekzod Fakhriddinov]

Nipun, ITSP is asking if I can get rid of  c= in ip4 x.x.x.x  field from the end of the invite I am sending  them.  I tried to create this profile and apply it on DIal-peer oubound but still sending them .  Is this possible to delete it ?  Their SBC sending SIP syntax error to my Invites, and as they think its because that c= field, I see invite include 2  same c=ip4 fields .

voice class sip-profiles 999
request INVITE sdp-header Video-Attribute remove
request INVITE sdp-header Video-Media modify "m=video(.*)" ""
request INVITE sdp-header Video-Bandwidth-Info remove
request REINVITE sdp-header Connection-Info remove
response 200 sdp-header Connection-Info remove
request INVITE sdp-header Connection-Info remove

[R0g22]

Connection data "c" is one of the mandatory SDP params per RFC 4566. Without that attribute, how does your ITSP plan to send RTP to the CUBE ?
Can you attach the INVITE that you see ? Every media description can have it's own connection data. That is completely legal again per the RFC.

[Bekzod Fakhriddinov]

As u can see I have 2 c= ...  fields in the Invite : 

Sent:

INVITE sip:19548219722@68.64.88.39:5060 SIP/2.0

Via: SIP/2.0/UDP MY_IP:5060;branch=z9hG4bK67EC081D39

Remote-Party-ID: "Bekzod F- 11273" <sip:2122205273@MY_IP>;party=calling;screen=yes;privacy=off

From: "Bekzod F- 11273" <sip:2122205273@MY_IP>;tag=2AD7EA8E-12C3

To: <sip:19548219722@UR_IP>

Date: Mon, 09 Apr 2018 10:58:42 GMT

Call-ID: 5226CDC5-3B3D11E8-B4A1B799-FD970943@MY_IP

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 1427283712-0000065536-0000229761-0219907082

User-Agent: Cisco-SIPGateway/IOS-15.4.3.M8

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1523285922

Contact: <sip:2122205273@MY_IP:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 68

Session-Expires:  1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 271

v=0

o=CiscoSystemsSIP-GW-UserAgent 4915 1841 IN IP4 MY_IP

s=SIP Call

c=IN IP4 MY_IP

t=0 0

m=audio 23890 RTP/AVP 0 101

c=IN IP4 MY_IP

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

c=IN IP4 MY_IP 

[R0g22]

v=0

o=CiscoSystemsSIP-GW-UserAgent 4915 1841 IN IP4 MY_IP

s=SIP Call

c=IN IP4 MY_IP

t=0 0

m=audio 23890 RTP/AVP 0 101

c=IN IP4 MY_IP

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

c=IN IP4 MY_IP

Was this copied correctly ? I see three "c" data in the SDP when you said that there are two.

[Bekzod Fakhriddinov]

U wa right, 3 c= lines. I could get rid of that last one, it was video not audio part , so i just replaced in sip profile request with these lines :

request INVITE sdp-header Video-Connection-Info remove
response 200 sdp-header Video-Connection-Info remove
request REINVITE sdp-header Video-Connection-Info remove

and ITSP also corrected on their side , I dont know what , but now there is audio traffic. 

Thank you for your response .

[R0g22]

Yes. Everything is correct. Like I mentioned, it is completely legal and RFC compliant to have connection data at session level and per media param. If both are present, than media param takes precedence.

[Bekzod Fakhriddinov]

HI Nipun ,  I have URI  rule on inbound dial-peer , uri is pointing to 3 ITSP servers, incoming call is coming and looks good but on the debug I see : 

cube uri vuri_compare_ip_address:IP Addresses are [ my public ip ,  my cucm ip ]NOT Equal

Is it some problem? I can't find this error in the internet.