Need help connecting to to SIP ITSP

shh5455 · ‎05-16-2010

Trying to connect to a SIP ITSP using CUCM 7.1.3 and a voice gateway running CUBE. When I dial out I can get the call to ring my cell phone, but when I answer I never get audio cut through. The IP phone doesn't appear to know that it answered as it keeps ringing. If I hang up the IP phone first, the cell phone will hang up. If I hang up the cell phone first the IP phone gets a reorder tone. So it almost seems like the communication is one-way. All devices are behind a NAT gateway.

Here is a 'debug ccsip messages' of an IP phone initiating and the cell phone answering.

*Mar 2 16:24:48.117: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2142884383@sip3.voipvoip.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.51:5060;branch=z9hG4bK191FBE
Remote-Party-ID: <sip:5557074097@10.0.1.51>;party=calling;screen=yes;privacy=off
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>
Date: Sat, 02 Mar 2002 16:24:48 GMT
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 2160533066-2488336831-134222081-167800370
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1015086288
Contact: <sip:5557074097@10.0.1.51:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0

*Mar 2 16:24:48.509: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.1.51:5060;branch=z9hG4bK191FBE;rport=55075;received=99.148.165.184
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.6097
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="voipvoip.com", nonce="4bf052c0734b1bea87a016d44450592c25ad109b"
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=13218 req_src_ip=99.148.165.184 req_src_port=55075 in_uri=sip:2142884383@sip3.voipvoip.com:5060 out_uri=sip:2142884383@sip3.voipvoip.com:5060 via_cnt==1"

*Mar 2 16:24:48.509: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:2142884383@sip3.voipvoip.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.51:5060;branch=z9hG4bK191FBE
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.6097
Date: Sat, 02 Mar 2002 16:24:48 GMT
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

*Mar 2 16:24:48.509: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2142884383@sip3.voipvoip.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.51:5060;branch=z9hG4bK1AA7B
Remote-Party-ID: <sip:5557074097@10.0.1.51>;party=calling;screen=yes;privacy=off
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>
Date: Sat, 02 Mar 2002 16:24:48 GMT
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 2160533066-2488336831-134222081-167800370
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1015086288
Contact: <sip:5557074097@10.0.1.51:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="5557074097",realm="voipvoip.com",uri="sip:2142884383@sip3.voipvoip.com:5060",response="cac47a6c8f8a7341ef171836477c65ac",nonce="4bf052c0734b1bea87a016d44450592c25ad109b",algorithm=md5
Content-Length: 0

CUBE#una
*Mar 2 16:24:48.745: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.0.1.51:5060;branch=z9hG4bK1AA7B;rport=55075;received=99.148.165.184
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
CSeq: 102 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=13219 req_src_ip=99.148.165.184 req_src_port=55075 in_uri=sip:2142884383@sip3.voipvoip.com:5060 out_uri=sip:VoIP$12142884383@sipgw.voipvoip.com:5960 via_cnt==1"

*Mar 2 16:24:53.853: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.51:5060;rport=55075;received=99.148.165.184;branch=z9hG4bK1AA7B
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>;tag=gK0bc7b4ef
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
CSeq: 102 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=8ABF900-1D9D>
CUBE#un alle: <sip:69.90.209.57:5060;nat=yes;ftag=8ABF900-1D9D;lr=on>
Contact: <sip:12142884383@208.70.13.195:5060>
Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,OPTIONS,MESSAGE,PUBLISH
Require: 100rel
RSeq: 11837
Content-Length: 261
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 32760 25378 IN IP4 208.70.13.195
s=SIP Media Capabilities
c=IN IP4 208.70.13.194
t=0 0
m=audio 27902 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

*Mar 2 16:24:54.353: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.51:5060;rport=55075;received=99.148.165.184;branch=z9hG4bK1AA7B
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>;tag=gK0bc7b4ef
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
CSeq: 102 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=8ABF900-1D9D>
Record-Route: <sip:69.90.209.57:5060
CUBE#un all;nat=yes;ftag=8ABF900-1D9D;lr=on>
Contact: <sip:12142884383@208.70.13.195:5060>
Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,OPTIONS,MESSAGE,PUBLISH
Require: 100rel
RSeq: 11837
Content-Length: 261
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 32760 25378 IN IP4 208.70.13.195
s=SIP Media Capabilities
c=IN IP4 208.70.13.194
t=0 0
m=audio 27902 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

*Mar 2 16:24:55.337: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.51:5060;rport=55075;received=99.148.165.184;branch=z9hG4bK1AA7B
From: <sip:5557074097@sip3.voipvoip.com>;tag=8ABF900-1D9D
To: <sip:2142884383@sip3.voipvoip.com>;tag=gK0bc7b4ef
Call-ID: D949C7CB-2D3011D6-8025C297-C23B33EC@10.0.1.51
CSeq: 102 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=8ABF900-1D9D>

Here's the configuration:

version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CUBE
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$Y/ur$GQJO8LNlQOlDxW8K3HJFY1
!
no aaa new-model
memory-size iomem 5
clock timezone CST -6
ip cef
!
!
!
!
ip name-server 10.0.1.11
ip name-server 4.2.2.2
multilink bundle-name authenticated
!
!
!
voice service voip
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
!
!
!
!
!
!
!
!
!
!
!
!
!
!
archive
log config
hidekeys
!
!
!
!
!
!
interface FastEthernet0/0
ip address 10.0.1.51 255.255.255.0
speed 100
full-duplex
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.0.1.51
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.0.1.1
!
ip http server
!
!
!
!
control-plane
!
!
!
!
!
!
!
dial-peer voice 2 voip
destination-pattern [2-9].........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
!
dial-peer voice 5 voip
description incoming dialplan
destination-pattern 5557074097
voice-class codec 1
session protocol sipv2
session target ipv4:10.0.1.13
incoming called-number .
dtmf-relay rtp-nte
!
!
sip-ua
authentication username 5557074097 password 7 <snip>
retry invite 2
retry bye 2
retry cancel 2
registrar dns:sip3.voipvoip.com expires 3600
sip-server dns:sip3.voipvoip.com
g729-annexb override
!
!
gatekeeper
shutdown

paolo bevilacqua · ‎05-17-2010

Recommend you eliminate NAT and have CUBE face the internet directy without any intermediate device. Otherwise, you will spend a lot of time and effort getting it to work, if it ever will.

shh5455 · ‎05-17-2010

I wish I could, but my DSL router isn't capable of running CUBE features and my ITSP requires authentication. My understanding is CUCM cannot do this yet natively. Can you tell from the SIP messages what the issue is? Thanks for the reply.

jaime.garcia · ‎05-17-2010

I just went through a huge migration with PAETEC from replacing all Data T1s and PRIs with SIP Trunks. We were having similar issues and it turned out to be the follwoing.

Make sure your Transcoders are properly configured in CUCM. I would also check with the ITSP on how digits they are passing to you. Make sure the correct about of digits are configured on your Gateway through CUCM.

Jaime

Felipe Garrido · ‎05-17-2010

The CUBE is not responding to the 183 with a PRACK. The "Require: 100rel" header in the 183 indicates that the CUBE should use reliable provisional responses.

What version of IOS is the CUBE running? Does the same problem occur for inbound calls? Is CUCM integrated via H.323 or SIP to the CUBE?

This could be caused by CSCsu50869.

shh5455 · ‎05-17-2010

Running 12.4(15)T13. The gateway integrates with CUCM via H.323.

Steven

Felipe Garrido · ‎05-18-2010

CSCsu50869 is not resolved in 12.4(15)T13. Would you be willing to upgrade to 12.4(22)T4 or 12.4(24)T3 to get the fix? You may also want to reference the Interop page to see if there is a tested configuration for the SIP SP being used.

http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html

shh5455 · ‎05-18-2010

I tried upgrading to 12.4(24)T3 which is the latest in the T train. I started getting one way audio (IP phone -> cell phone) but no ringback or anything. After looking at those SIP configs you linked to I decided that they are so varied by the carrier that it's not very likely my small carrier is going to be able to help me. The one way audio was so delayed anyway that it would be unusable even if it was 2way. So I think I'm just going to scrap the project.

Thanks for your help anyway!

ken.kopp · ‎06-07-2010

I have been looking for a solution to what I believe may be problems caused buy what I think is a issue in your config as well.

That is, you are using an internet TSP as we are, yet your bound voip GW is set to your internal interface.

We have been experencing seemingly random one-way audio.

Several days of packet capture revealed that while the SIP traffic was as expected the RTP was not.

When a call is placed & built SIP goes over without a problem, however, the RTP is sent from us with our GW internal (non-Internet-routable) address as the source address in the header. I believe the cure may be to bind the 323 GW address to the external interface and & hence the Internet routable address. Then of course point the CM at the new address.

Just theory at this point, I am still looking for conformation of the correct interface & address when using an Internet based TSP.

I hope this helps in some way!