SIP Trunk Configuration on SBX IP

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Transcript SIP Trunk Configuration on SBX IP

VoIP made simple.
Welcome to Vertical’s
SBX IP
Technical Webinar
Objective
To provide Vertical dealers with the necessary tools,
information and experience essential to the successful
deployment and implementation of the SBX IP 320
Topic
KSU Configuration for Session Initiation Protocol (SIP)
Trunks.
Tools You Will Need
1.
Laptop
2.
Network Diagram
3.
Ethernet Cables (Straight-through and Crossover)
4.
Ethernet Hub
5.
Wireshark (Free Packet Capture Software)
Being Prepared Will Save You Time
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Requirements
1.
Static Public IP Address for the VOIB (cannot be shared)
2.
Direct/Routed Internet Access or 1-to-1 NAT Capable Device
3.
Certified SIP Carrier And Account Information (Certified Carrier Documentation Available On
VConnect)
4.
VOIB
5.
SBX PCAdmin Software
•
•
Port Forwarding Is NOT Supported For VOIB
PAT Is NOT Supported For VOIB
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IP Address Matrix
Address Class
1st Decimal Range
Network/Host ID**
Default Subnet Mask
A
1 - 126*
N.H.H.H
255.0.0.0 or /8
B
128 - 191
N.N.H.H
255.255.0.0 or /16
C
192 - 223
N.N.N.H
255.255.255.0 or /24
D
224 - 239
Multicast
E
240 - 254
Experimental
** N = Network ID
H = Host ID
Private IP Addresses:
Class A 10.0.0.0 - 10.255.255.255
Class B 172.16.0.0 – 172.31.255.255
Class C 192.168.0.0 –192.168.255.255
* 127.0.0.0 is reserved for loopback diagnostic functions
0.0.0.0 is a “special case” available for use as a broadcast address.
Static IP Addresses are manually assigned
and do not change.
Dynamic IP Addresses are assigned via DHCP
and may change from day to day.
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Direct Internet Access Diagram
Data Network
Carrier
Internet
Router
Hub or
Switch
KSU
VOIB With A
Static Public IP
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1-to-1 NAT Capable Device Internet Access Diagram
Data Network
1-to-1 NAT Capable
Carrier
Internet
Router
Hub or
Switch
KSU
1-to-1 NAT Statement
Mapping Static Public IP
To Private IP Assigned to VOIB
Do Not Confuse This With
Port Forwarding/Port Triggering
*Additional Programming on VOIB
VOIB With A
Static Private IP
Common Network Diagram Using Multiple Public IP’s
Data Network
Carrier
Internet
Hub or
Switch
Router
Routes to Multiple Public IP Addresses
Inside the Network
KSU
VOIB With A
Static Public IP
What Is SIP
The Session Initiation Protocol (SIP) is a signaling protocol used for establishing sessions in an IP network. A
session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. SIP
is developed purely as a mechanism to establish sessions, it does not know about the details of a session,
it just initiates, terminates and modifies sessions.
Over the last couple of years, the Voice over IP community has adopted SIP as its protocol of choice for signaling.
SIP is an RFC standard (RFC 3261) from the Internet Engineering Task Force (IETF), the body responsible for
administering and developing the mechanisms that comprise the Internet. SIP is still evolving and being extended as
technology matures and SIP products are socialized in the marketplace.
SIP Messages
SIP Is A Text-based Protocol. The Client Makes Requests and the Server Returns
Answers to Client Requests Using Two Types of Messages. Requests (Methods)
And Answers (State Codes).

1xx Provisional/Informational Response – Request Received and Processing
*A Server sends a 1xx response if it expects to take more than 200 ms to obtain a final response

2xx Success – The Action Was Successfully Received Understood Accepted

3xx Redirection – Further Action Needs To Be Taken To Complete Request

4xx Method/Client Error

5xx Server Failure/Error

6xx Global Failure/Error
SIP Methods
The Initial Line/Request Line Is Most Important Part of SIP Request. It Contains
the Method Name, Request URI and SIP Protocol Version. There are Six Basic
Methods (RFC 254) for Client Requests

INVITE:

ACK: Confirm Session Establishment

OPTION:

BYE:

CANCEL:
Cancel a Pending Request

REGISTER:
Register the User Agent
Invite a User or a Service to a New Session/Modify Session
Request Information About the Capabilities of a Server
End of a Session
SIP Call Setup
We go off-hook and dial a number. Authentication/registration with the
Carrier
Occurs and is accepted.
Call is processed by carrier and returns ring to us.
Call is answered and connected.
Conversation takes place.
We end the call and it is disconnected and
the line cleared.
SIP Error Messages

4xx Method Failures/Client Error - Generally Authentication Failure




- Authentication Failure
- Call Rejected
- Authentication Failure
5xx Server Failure/Server Errors - Server Failed to fulfill a Valid Request



401 Unauthorized
402 Payment Required
403 Forbidden
503 Service Unavailable
504 Gateway Timeout
- SIP Server May Be Down
- SIP Server May Be Down
6xx Global Failure/Global Errors – Request Cannot Be Fulfilled at Any Server


600 Busy Everywhere
606 Does Not Exist Anywhere
- User Busy
- Unallocated Number
*Many 4xx Errors Report On The Station Display
SIP Call Trace Capture – Failed Call
[SIP-CMD] INVITE sip:6024050155
4.79.212.235;user=phone
From:<sip:1234567890
4.79.212.235>
To:<sip:6024050155
4.79.212.235;user=phone>
Contact:sip:
172.19.13.242:5060
010426 C>10 04, D5 B5 09 1D 3C 73 69 70 3A 31 32 33 34 35 36 37 38 39 30 40 34 2E 37 39 2E 32 31 32 2E 32 33 35 3E
0A
“”Truncated””
[SIP-EVT] SIP_CALLFAIL_RESP_MSG
IE_SIP_RESPONSE_CODE:403
IE_SIP_CONTACT:sip:172.19.13.242:5060
DEMONSTRATION
www.sbxip320.com
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Contact Information
•Training Available
*Vertical University (Web Based)
https://university.vertical.com
•Software and Documents Available
*VConnect
http://vconnect.vertical.com
•Technical Support
**Online
http://view.vertical.com
**Phone
1-877-Vertical
* Requires Login
** Requires Tech Number
Thank You
Questions & Answers
www.sbxip320.com
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