03-21-2018 11:58 AM - edited 03-17-2019 12:28 PM
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
Solved! Go to Solution.
03-22-2018 10:30 AM
03-22-2018 10:48 AM
Nipun, great explanation, thank you , you are the best qualified, best patient engineer.
03-22-2018 12:16 PM
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 ?
03-22-2018 12:39 PM
03-26-2018 07:23 AM - edited 03-26-2018 07:38 AM
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?
03-26-2018 08:39 AM
03-28-2018 10:45 AM
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
03-28-2018 10:56 AM
04-09-2018 11:36 AM
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
04-09-2018 01:03 PM
04-09-2018 01:05 PM
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
04-09-2018 01:21 PM
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.
04-09-2018 02:10 PM
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 .
04-09-2018 02:30 PM
04-11-2018 11:10 AM
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.
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