Third party SIP trunk problem

[msk69]

I have two sites having CM 7.1, System and Session Manager release 7.1

Main Site= CM + System Manager + Session Manager
DR Site= CM + Session Manager + Media Server+ SBC rls 7.2

I have hooked up a SIP trunk to DR site SBC. When i call from main site to any cell number or Land line number through DR SIP trunk i get dead air, no voice in or out.
Can somebody help from where to start troubleshooting. I check by logging a Soft phone user from Main and DR site. I also checked with hard phone from both site.
The SIP trunk provider saying that we are sending double invite. However when i trace SM and SBC it look quite OK.
Waiting for a needful response.

 

[kyle555]

I'd look at a sip trace with the SDP on the inside of the SBC. Maybe you don't have routes or ports open between the audio endpoints.

 

[msk69]

Hi Kyle, I have checked the ports twice, 5060 and 5061 are both open. Below is the call trace on session manager, an extension is calling outside to a cell number. For safety purpose i have hide a part of IPs. Please note that the SBC is on DR site only. My call to SIP trunk is going out through Main site session Manager. The SIP trunk is hooked on DR site with SBC.

Trace
=====

FISM01 FIDRSBC01 SM100 ─────────────┬───────────┬───────────┬───────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────14:11:21.495 │──INVITE──►│ │ │ (2) T:03218706393 F:5924 U:03218706393 Priginating
14:11:21.499 │◄──Trying──│ │ │ (2) 100 Trying
14:11:21.557 │ │──INVITE──►│ │ (2) T:03218706393 F:36490050 U:03218706393 P:terminating
14:11:21.566 │ │◄──Trying──│ │ (2) 100 Trying
14:11:23.450 │ │◄──OPTIONS─│ │ (3) sip:customer.com
14:11:23.454 │ │──200 OK──►│ │ (3) 200 OK (OPTIONS)
14:11:25.476 │ │◄──Ringing─│ │ (2) 180 Ringing
14:11:25.506 │◄──Ringing─│ │ │ (2) 180 Ringing
14:11:27.890 │ │◄──200 OK──│ │ (2) 200 OK (INVITE)
14:11:27.909 │◄──200 OK──│ │ │ (2) 200 OK (INVITE)
14:11:27.930 │────ACK───►│ │ │ (2) sip:192.168.x.xx:5060
14:11:27.933 │ │────ACK───►│ │ (2) sip:192.168.x.xx:5060
14:11:33.450 │ │◄──OPTIONS─│ │ (5) sip:customer.com
14:11:33.454 │ │──200 OK──►│ │ (5) 200 OK (OPTIONS)
14:11:37.388 │ │◄────BYE───│ │ (2) sip:5924@192.168.xx.xx:55299
14:11:37.391 │◄────BYE───│ │ │ (2) sip:5924@192.168.xx.xx:55299
14:11:37.724 │──200 OK──►│ │ │ (2) 200 OK (BYE)
14:11:37.731 │ │──200 OK──►│ │ (2) 200 OK (BYE)
14:11:43.450 │ │◄──OPTIONS─│ │ (7) sip:customer.com
14:11:43.454 │ │──200 OK──►│ │ (7) 200 OK (OPTIONS)
14:11:43.715 │ │◄──OPTIONS─│ │ (8) sip:customer.com
14:11:43.718 │ │──200 OK──►│ │ (8) 200 OK (OPTIONS)
14:11:45.265 │ │──OPTIONS─►│ │ (9) sip:192.168.0.190
14:11:45.319 │ │◄──200 OK──│ │ (9) 200 OK (OPTIONS)

 

[kyle555]

Yeah, but you want to expand upon the INVITE, the RINGING, the 200 and ACK and see which had SDP and where they were instructing the audio to be sent. That's the packet flow you want to debug

 

[msk69]

I am trying but since i am not very fond of Trace stuff so i got stuck here. Can somebody help how to troubleshoot or from where to start?

 

[msk69]

Hi Kyle,
Below is the detail capture

SIP/2.0 180 Ringing
==================== ¦
¦From: <sip:36490050@customer.com>;tag=c17fb6e9-075d-4a67-b7df-167fa8822b1f ¦
¦To: <sip:03218706393@customer.com>;tag=7739e8c2-CC-32 ¦
¦CSeq: 2 INVITE ¦
¦Call-ID: d52dc27d-4c23-49a7-9ac5-91761504e855 ¦
¦Contact: <sip:192.168.0.190:5060;user=phone;transport=tcp> ¦
¦Record-Route: <sip:FIDRSM01@192.168.0.22;transport=tcp;lr;av-asset-uid=b0d2682> ¦
¦Record-Route: <sip:127.0.0.2:15060;transport=tcp;ibmsid=local.1592047744020_194568_194567;lr> ¦
¦Record-Route: <sip:FIDRSM01@192.168.0.22;transport=tcp;lr;av-asset-uid=b0d2682> ¦
¦Record-Route: <sip:FISM01@192.168.104.73;transport=tcp;lr;av-asset-uid=3f82cecb> ¦
¦Record-Route: <sip:127.0.0.2:15060;transport=tcp;ibmsid=local.1575245299315_8866722_8868072;lr> ¦
¦Record-Route: <sip:FISM01@192.168.104.73;transport=tcp;lr;av-asset-uid=3f82cecb> ¦
¦Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER ¦
¦Supported: replaces ¦
¦Via: SIP/2.0/TCP 192.168.0.22;branch=z9hG4bK812692229121672-AP;ft=3 ¦
Via: SIP/2.0/TCP 127.0.0.2:15060;rport=30547;ibmsid=local.1592047744020_194569_194568;branch=z9hG4bK812692229121672 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport;ibmsid=local.1592047744020_194568_194567;branch=z9hG4bK163432536018617 ¦
¦Via: SIP/2.0/TCP 192.168.0.22;branch=z9hG4bK439241414083070-AP-AP;ft=20;received=192.168.0.22;rport=46325 ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK439241414083070-AP;ft=7 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport=31843;ibmsid=local.1575245299315_8866723_8868073;branch=z9hG4bK439241414083070 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport;ibmsid=local.1575245299315_8866722_8868072;branch=z9hG4bK24470782121411 ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK0ccd4f3a-7d81-456c-a997-16f16025c4de-AP;ft=6393;received=192.168.104.73;rport=32165 ¦
¦Via: SIP/2.0/TCP 192.168.108.90:62292;branch=z9hG4bK0ccd4f3a-7d81-456c-a997-16f16025c4de ¦
¦Content-Type: application/sdp ¦
¦Content-Length: 216 ¦
¦ ¦
¦v=0 ¦
¦o=HuaweiSoftX3000 40974229 40974229 IN IP4 202.125.137.5 ¦
¦s=Sip ¦
¦c=IN IP4 202.125.137.5 ¦
¦t=0 0 ¦
¦m=audio 35028 RTP/AVP 8 120 ¦
¦a=rtpmap:8 PCMA/8000 ¦
¦a=rtpmap:120 telephone-event/8000 ¦
¦a=fmtp:120 0-15 ¦
¦a=ptime:20 ¦
+--------------

INVITE
====== INVITE sip:03218706393@customer.com SIP/2.0 ¦
¦Route: <sip:192.168.0.22;transport=tcp;lr;phase=originating;seq=explicit;m-type=audio;remote-ure-route=true> ¦
¦Max-Breadth: 60 ¦
¦P-Conference: OutdialPrompt=false, FeedbackPrompts=false, UCCP=true, Video=true, WebCollaboration=true ¦
¦Accept-Language: en ¦
¦Contact: <sip:5924@192.168.108.90:62292;transport=tcp;gsid=510026f0-b078-11ea-8c5a-005056b61ca3;av-iptol>;+sip.instance="<urn:uuid:8fb1b0a3-f651-4ca7-b869-e6d9d¦
¦7e3d0f3>" ¦
¦Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, NOTIFY, MESSAGE, REFER, INFO, PUBLISH, UPDATE ¦
¦Supported: eventlist, outbound, replaces ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK586398777241356-AP;ft=7 ¦
¦Via: SIP/2.0/TCP127.0.0.2:15060;rport=31843;ibmsid=local.1575245299315_8867159_8868509;branch=z9hG4bK586398777241356 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport;ibmsid=local.1575245299315_8867158_8868508;branch=z9hG4bK397869758604990 ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK7223019a-2f31-4ffc-81de-a328ce27544c-AP;ft=6393;received=192.168.104.73;rport=32165 ¦
¦Via: SIP/2.0/TCP 192.168.108.90:62292;branch=z9hG4bK7223019a-2f31-4ffc-81de-a328ce27544c ¦
Record-Route: <sip:FISM01@192.168.104.73;transport=tcp;lr;av-asset-uid=3f82cecb> ¦
¦Record-Route: <sip:127.0.0.2:15060;transport=tcp;ibmsid=local.1575245299315_8867158_8868508;lr> ¦
¦Record-Route: <sip:FISM01@192.168.104.73;transport=tcp;lr;av-asset-uid=3f82cecb> ¦
¦User-Agent: Avaya Communicator/3.0 (3.7.4.22.1; Avaya CSDK; Microsoft Windows NT 6.2.9200.0) AVAYA-SM-7.1.3.0.713014 ¦
¦P-Charging-Vector: icid-value="510026f0-b078-11ea-8c5a-005056b61ca3" ¦
¦P-Asserted-Identity: "khan, Muhammad" <sip:5924@customer.com> ¦
¦P-AV-Message-Id: 1_1 ¦
¦From: <sip:5924@customer.com>;tag=ff85e2c8-cdd5-41df-b688-765e75ab7c05 ¦
¦To: <sip:03218706393@customer.com> ¦
¦Call-ID: 71d41f4a-48d7-4b5b-b756-cd883edced06 ¦
¦Max-Forwards: 66 ¦
¦CSeq: 2 INVITE ¦
¦Content-Type: application/sdp ¦
¦Content-Length: 558 ¦
¦Av-Global-Session-ID: 510026f0-b078-11ea-8c5a-005056b61ca3 ¦
¦P-Site: SM;smgr=192.168.104.71;origloc=163840;origmedialoc=163840;orighomeloc=163840;termloc=163840;termsigloc=163840;terment=229377 ¦
¦P-Location: SM;origlocname="FalconOfficeI";origmedialocname="FalconOfficeI";orighomelocname="FalconOfficeI";termlocname="FalconOfficeI";termsiglocname="FalconOf¦
¦ficeI";smaccounting="false" ¦
¦ ¦
¦v=0 ¦
¦o=sip:5924@192.168.108.90 3 2 IN IP4 192.168.108.90 ¦
¦s=- ¦
¦c=IN IP4 192.168.108.90 ¦
¦b=TIAS:64000 ¦
¦t=0 0 ¦
¦a=activetalker:1 ¦
¦m=audio 5010 RTP/AVP 116 9 8 0 110 18 120 ¦
¦a=sendrecv




ACKNOWLEDGE
===========

¦ACK sip:192.168.0.190:5060;transport=tcp;user=phone SIP/2.0 ¦
¦P-Location: SM;origlocname="FalconOfficeI";origmedialocname="FalconOfficeI";orighomelocname="FalconOfficeI";termlocname="FalconOfficeI";termsiglocname="FalconOf¦
¦ficeI";termmedialocname="FalconOfficeI";smaccounting="true" ¦
1¦User-Agent: Avaya Communicator/3.0 (3.7.4.22.1;Avaya CSDK; Microsoft Windows NT 6.2.9200.0) AVAYA-SM-7.1.3.0.713014 ¦
¦Av-Global-Session-ID: 9570d250-b076-11ea-8c5a-005056b61ca3 ¦
¦Via: SIP/2.0/TCP 192.168.0.22;branch=z9hG4bK289427346454919-AP;ft=3 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport=30547;ibmsid=local.1592047744020_194568_194567;branch=z9hG4bK289427346454919 ¦
¦Via: SIP/2.0/TCP 192.168.0.22;branch=z9hG4bK429291883119160-AP-AP;ft=20;received=192.168.0.22;rport=46325 ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK429291883119160-AP;ft=7 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport=31843;ibmsid=local.1575245299315_8866722_8868072;branch=z9hG4bK429291883119160 ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK1287576c-3c87-4f08-8f0b-9a8a4ff6c139-AP;ft=6393;received=192.168.104.73;rport=32165 ¦
¦Via: SIP/2.0/TCP 192.168.108.90:62292;branch=z9hG4bK1287576c-3c87-4f08-8f0b-9a8a4ff6c139 ¦
¦Supported: eventlist, outbound, replaces ¦
¦Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, NOTIFY, MESSAGE, REFER, INFO, PUBLISH, UPDATE ¦
¦Contact: <sip:5924@192.168.108.90:62292;transport=tcp;gsid=9570d250-b076-11ea-8c5a-005056b61ca3>;+sip.instance="<urn:uuid:8fb1b0a3-f651-4ca7-b869-e6d9d7e3d0f3>"¦
¦ ¦
¦From: <sip:36490050@customer.com>;tag=c17fb6e9-075d-4a67-b7df-167fa8822b1f ¦
¦To: <sip:03218706393@customer.com>;tag=7739e8c2-CC-32 ¦
¦Call-ID: d52dc27d-4c23-49a7-9ac5-91761504e855 ¦
¦Max-Forwards: 64 ¦
¦CSeq: 2 ACK


200 ok
=======
SIP/2.0 200 OK ¦
¦From: <sip:36490050@customer.com>;tag=c17fb6e9-075d-4a67-b7df-167fa8822b1f ¦
¦To: <sip:03218706393@customer.com>;tag=7739e8c2-CC-32 ¦
¦CSeq: 2 INVITE ¦
¦Call-ID: d52dc27d-4c23-49a7-9ac5-91761504e855 ¦
¦Contact: <sip:192.168.0.190:5060;user=phone;transport=tcp> ¦
¦Record-Route: <sip:FIDRSM01@192.168.0.22;transport=tcp;lr;av-asset-uid=b0d2682> ¦
¦Record-Route: <sip:127.0.0.2:15060;transport=tcp;ibmsid=local.1592047744020_194568_194567;lr> ¦
¦Record-Route: <sip:FIDRSM01@192.168.0.22;transport=tcp;lr;av-asset-uid=b0d2682> ¦
¦Record-Route: <sip:FISM01@192.168.104.73;transport=tcp;lr;av-asset-uid=3f82cecb> ¦
¦Record-Route: <sip:127.0.0.2:15060;transport=tcp;ibmsid=local.1575245299315_8866722_8868072;lr> ¦
¦Record-Route: <sip:FISM01@192.168.104.73;transport=tcp;lr;av-asset-uid=3f82cecb> ¦
¦Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER ¦
¦Supported: replaces ¦
¦Via: SIP/2.0/TCP 192.168.0.22;branch=z9hG4bK812692229121672-AP;ft=3 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport=30547;ibmsid=local.1592047744020_194569_194568;branch=z9hG4bK812692229121672 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport;ibmsid=local.1592047744020_194568_194567;branch=z9hG4bK163432536018617 ¦
¦Via: SIP/2.0/TCP 192.168.0.22;branch=z9hG4bK439241414083070-AP-AP;ft=20;received=192.168.0.22;rport=46325 ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK439241414083070-AP;ft=7 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport=31843;ibmsid=local.1575245299315_8866723_8868073;branch=z9hG4bK439241414083070 ¦
¦Via: SIP/2.0/TCP 127.0.0.2:15060;rport;ibmsid=local.1575245299315_8866722_8868072;branch=z9hG4bK24470782121411 ¦
¦Via: SIP/2.0/TCP 192.168.104.73;branch=z9hG4bK0ccd4f3a-7d81-456c-a997-16f16025c4de-AP;ft=6393;received=192.168.104.73;rport=32165 ¦
¦Via: SIP/2.0/TCP 192.168.108.90:62292;branch=z9hG4bK0ccd4f3a-7d81-456c-a997-16f16025c4de ¦
¦Content-Type: application/sdp ¦
¦Content-Length: 216 ¦
¦ ¦
¦v=0 ¦
¦o=HuaweiSoftX3000 40974229 40974230 IN IP4 202.125.137.5 ¦
¦s=Sip ¦
¦c=IN IP4 202.125.137.5 ¦
¦t=0 0 ¦
¦m=audio 35028 RTP/AVP 8 120 ¦
¦a=rtpmap:8 PCMA/8000 ¦
¦a=rtpmap:120 telephone-event/8000 ¦
¦a=fmtp:120 0-15 ¦
¦a=ptime:20 ¦
+----------------------------------------------------------------------------------------------------------------------------------------------------------------+


Any help now??

 

[kyle555]

well, your problem is pretty clear. You`re asking a 202. IP and a 192. IP to talk. That needs to be translated somewhere.

You`re telling the Huawei switch to send packets to 192.168.108.90. How would it know to do that?

What IPs do you have setup for your carrier, for your B1, A1, and SM?

 

[msk69]

A1= 192.168.0.190 (connected with session manager, It is SBC IP))
A2= 202.125.137.5 (SIP Service provider NIC IP)
M1= 192.168.0.183
B1= Not connected/ Not used

VPN IP= 192.168.108.90 ( i am connected with customer network using VPN. A softphone is installed on my LAPTOP and
i am testing by dialing a cell no through this softphone. The dialout call goes out through SIP trunk.

Any more advise?

 

[kyle555]

The originator of the SDP for the 202.125.137.5 is the Huawei switch

You might choose to reach it thru A2, but 202.125.137.5 is not the IP of the A2 interface. What is it?

 

[kyle555]

Normally you use B1 for public side. The IPs have a public IP override, so if you have Public IP 101.102.103.104 on your WAN router and 192.168.10.10 on B1, that will make you send SDP to the carrier with c=line of 101.102.103.104. Just make sure the rtp ports of the media interface are forwarded toward the b1 interface too.

 

[msk69]

Hi Kyle, I raised a SR. Avaya is looking into it.

 

[msk69]

Hi Kyle,
Problem resolved. The problem was in communication manager UDP settings. After increasing port range the problem resolved.