CUCM direct SIP trunk to SIP Provider

Answered Question
Aug 11th, 2010

I was trying to setup a SIP trunk using CUBE to Bandwidth.com and while it was working on inbound, with an h.323 connection to CUCM, calls were getting dropped due to the lack of a high enough CUBE version.  Since I can not upgrade CUBE due to hardware limitations (3745 router) I am trying to terminate the trunk directly into CUCM.  Two issues are arising.  With inbound calls they don't connect because there is a 404 error, apparently the DID number is not found.  I'm receiving 10 digits and I have the DID programmed as a DN and assigned to an IP phone so it ought to ring directly but calls from the PSTN are not coming inbound.  Second issue is outbund calls are one-way audio.  The person on the other end (PSTN) can hear me but I can't hear anything.  I thought NAT at first but that doesn't appear to be the case.  Im including 2 SIP traces from Bandwidth.com, one for inbound calls where the DID can't be found (404 error) and the other demonstrating the 1-way audio on outbound calls.  It appears that what is causing the 1-way audio is that the IP phone itself (7970G, SCCP) is rewriting the IP address, so the phone's IP is coming through in the SDP instead of the routable IP that it should be.  So the issue seems more on the phone, not sure why it is re-writing the IP but that seems to be the root of the issue for 1-way audio.  Any thoughts?

Outbound Call:

U 2010/08/12 02:58:28.432308 96.xx.xx.xx:5060 -> 216.82.224.202:5060
INVITE sip:[email protected]:5060 SIP/2.0.
Via: SIP/2.0/UDP 96.56.78.170:5060;branch=z9hG4bK425be467d1.
From: "Ronan McGurn" <sip:[email protected]>;tag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567.
To: <sip:[email protected]>.
Date: Fri, 13 Aug 2010 06:02:39 GMT.
Call-ID: [email protected].
Supported: timer,resource-priority,replaces.
Min-SE:  1800.
User-Agent: Cisco-CUCM8.0.
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY.
CSeq: 101 INVITE.
Contact: <sip:[email protected]:5060>.
Expires: 180.
Allow-Events: presence.
Supported: X-cisco-srtp-fallback.
Supported: Geolocation.
Cisco-Guid: 1600171776-3326205951-0000000016-0033558700.
Session-Expires:  1800.
P-Asserted-Identity: "Ronan McGurn" <sip:[email protected]>.
Remote-Party-ID: "Ronan McGurn" <sip:[email protected]>;party=calling;screen=yes;privacy=off.
Max-Forwards: 70.
Content-Length: 0.
.

U 2010/08/12 02:58:28.434308 216.82.224.202:5060 -> 96.xx.xx.xx:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 96.56.78.170:5060;branch=z9hG4bK425be467d1.
From: "Ronan McGurn" <sip:[email protected]>;tag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567.
To: <sip:[email protected]>.
Call-ID: [email protected].
CSeq: 101 INVITE.
Server: Bandwidth.com TRM (bw7.gold.13).
Content-Length: 0.
.


U 2010/08/12 02:58:29.678146 216.82.224.202:5060 -> 96.xx.xx.xx:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 96.56.78.170:5060;branch=z9hG4bK425be467d1.
From: "Ronan McGurn" <sip:[email protected]>;tag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567.
To: <sip:[email protected]>;tag=gK05b0c9e9.
Call-ID: [email protected].
CSeq: 101 INVITE.
Record-Route: <sip:216.82.224.202:5060;lr;ftag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567>.
Contact: <sip:[email protected]:5060>.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH.
Content-Length: 0.
.

U 2010/08/12 02:58:29.855618 216.82.224.202:5060 -> 96.xx.xx.xx:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 96.56.78.170:5060;branch=z9hG4bK425be467d1.
From: "Ronan McGurn" <sip:[email protected]>;tag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567.
To: <sip:[email protected]>;tag=gK05b0c9e9.
Call-ID: [email protected].
CSeq: 101 INVITE.
Record-Route: <sip:216.82.224.202:5060;lr;ftag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567>.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed.
Contact: <sip:[email protected]:5060>.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH.
Content-Length:  234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 16368 5111 IN IP4 67.231.0.72.
s=SIP Media Capabilities.
c=IN IP4 67.231.8.106.
t=0 0.
m=audio 12294 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/08/12 02:58:29.922160 96.xx.xx.xx:5060 -> 216.82.224.202:5060
ACK sip:[email protected]:5060 SIP/2.0.
Via: SIP/2.0/UDP 96.56.78.170:5060;branch=z9hG4bK436dcb7794.
From: "Ronan McGurn" <sip:[email protected]>;tag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567.
To: <sip:[email protected]>;tag=gK05b0c9e9.
Date: Fri, 13 Aug 2010 06:02:39 GMT.
Call-ID: [email protected].
Route: <sip:216.82.224.202:5060;lr;ftag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019567>.
Max-Forwards: 70.
CSeq: 101 ACK.
Allow-Events: presence.
Content-Type: application/sdp.
Content-Length: 213.
.
v=0.
o=CiscoSystemsCCM-SIP 2000 2 IN IP4 172.16.0.2.
s=SIP Call.
c=IN IP4 192.168.110.205.
t=0 0.
m=audio 32516 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

Where it says "192.168.110.205" it should be the routable IP, 96.xx.xx.xx.  Then there should be 2-way audio, just not sure how to make this happen.

-----------------------

This is the inbound trace information where it can't find the DID number, the 404 error.

U 2010/08/12 02:47:10.439854 216.82.224.202:5060 -> 96.xx.xx.xx:5060
INVITE sip:[email protected]:5060;transport=udp SIP/2.0.
Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460;vsf=AAAAABIAAAMLCgAKBwJ1BG4CFhgKGwIbARoJNDg->.
Record-Route: <sip:67.231.8.93;lr=on;ftag=VPSF506071629460>.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK3901.78353cf5.0.
Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bK3901.dc69e8e2.0.
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1257041398292.
From: "BANDWIDTH COM"  <sip:[email protected]>;tag=VPSF506071629460.
To: <sip:[email protected]:5060>.
Call-ID: [email protected].
CSeq: 1 INVITE.
Contact: <sip:[email protected]:5060;transport=udp>.
Max-Forwards: 67.
Content-Type: application/sdp.
Content-Length: 173.
.
v=0.
o=- 1281581230 1281581231 IN IP4 209.247.5.136.
s=-.
c=IN IP4 209.247.5.136.
t=0 0.
m=audio 61252 RTP/AVP 0 18 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 2010/08/12 02:47:10.453808 96.xx.xx.xx:5060 -> 216.82.224.202:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP  216.82.224.202;branch=z9hG4bK3901.78353cf5.0,SIP/2.0/UDP  67.231.8.93;branch=z9hG4bK3901.dc69e8e2.0,SIP/2.0/UDP  4.68.250.148:5060;branch=z9hG4bK506071629460-1257041398292.
From: "BANDWIDTH COM"  <sip:[email protected]>;tag=VPSF506071629460.
To: <sip:[email protected]:5060>.
Date: Fri, 13 Aug 2010 05:51:21 GMT.
Call-ID: [email protected].
CSeq: 1 INVITE.
Allow-Events: presence.
Content-Length: 0.
.


U 2010/08/12 02:47:10.455909 96.xx.xx.xx:5060 -> 216.82.224.202:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP  216.82.224.202;branch=z9hG4bK3901.78353cf5.0,SIP/2.0/UDP  67.231.8.93;branch=z9hG4bK3901.dc69e8e2.0,SIP/2.0/UDP  4.68.250.148:5060;branch=z9hG4bK506071629460-1257041398292.
From: "BANDWIDTH COM"  <sip:[email protected]>;tag=VPSF506071629460.
To: <sip:[email protected]:5060>;tag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019564.
Date: Fri, 13 Aug 2010 05:51:21 GMT.
Call-ID: [email protected].
CSeq: 1 INVITE.
Allow-Events: presence.
Reason: Q.850;cause=1.
Content-Length: 0.
.


U 2010/08/12 02:47:10.455985 216.82.224.202:5060 -> 96.xx.xx.xx:5060
ACK sip:[email protected]:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK3901.78353cf5.0.
From: "BANDWIDTH COM"  <sip:[email protected]>;tag=VPSF506071629460.
Call-ID: [email protected].
To: <sip:[email protected]:5060>;tag=6680d3de-3c8f-4a5d-a2c4-8f886b8fd3df-26019564.
CSeq: 1 ACK.
Max-Forwards: 70.
User-Agent: Bandwidth.com TRM (bw7.gold.13).
Content-Length: 0.

Thank you very much for any light anyone can shed on this issue.

CUCM is version 8.0.  Trying to set this up in a lab environment and take it from there.

I have this problem too.
0 votes
Correct Answer by srgudava about 6 years 4 months ago

Hi

For the out bound call I am not sure if there is a way to implement this with out a CUBE for Audio part. As you need to Nat all the RTP stream to Public and back to Private IP of the Phone.

For the inbound call the reason why the call is failing is that the inbound Invite came with

To: <sip:[email protected]:5060>.

When the Invite hits the call manager with this To header we will look in the call manager Device table if there is any Ipaddress:port with this in the Sip-trunk device table if not match we will go a head and send 404 Not found.

HTH

Sri Gudavalli

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Correct Answer
srgudava Thu, 08/12/2010 - 01:28

Hi

For the out bound call I am not sure if there is a way to implement this with out a CUBE for Audio part. As you need to Nat all the RTP stream to Public and back to Private IP of the Phone.

For the inbound call the reason why the call is failing is that the inbound Invite came with

To: <sip:[email protected]:5060>.

When the Invite hits the call manager with this To header we will look in the call manager Device table if there is any Ipaddress:port with this in the Sip-trunk device table if not match we will go a head and send 404 Not found.

HTH

Sri Gudavalli

fieryhail Thu, 08/12/2010 - 09:59

Thank you very much for your response.  I guess the situation is then I need to upgrade the 3745 router so that I can then upgrade the CUBE element.  Barring that or a third-party server or appliance I will not be able to bring a SIP trunk into CUCM.  I very much appreciate your time with this.  Thank you.

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