Presentation on theme: "Welcome to Verticals Technical Webinar SBX IP VoIP made simple."— Presentation transcript:
Welcome to Verticals Technical Webinar SBX IP VoIP made simple.
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.
3 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
4 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 VOIBPort Forwarding Is NOT Supported For VOIB PAT Is NOT Supported For VOIBPAT Is NOT Supported For VOIB
5 IP Address Matrix Address Class1st Decimal RangeNetwork/Host ID**Default Subnet Mask A *N.H.H.H or /8 B N.N.H.H or /16 C N.N.N.H or /24 D Multicast E Experimental ** N = Network IDH = Host ID Private IP Addresses: Class A Class B – Class C – * is reserved for loopback diagnostic functions 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.
6 Direct Internet Access Diagram KSU Hub or Switch Router VOIB With A Static Public IP Data Network Internet Carrier
1-to-1 NAT Capable Device Internet Access Diagram KSU Hub or Switch Router VOIB With A Static Private IP Data Network Internet Carrier 1-to-1 NAT Statement 1-to-1 NAT Statement Mapping Static Public IP Mapping Static Public IP To Private IP Assigned to VOIB To Private IP Assigned to VOIB Do Not Confuse This With Port Forwarding/Port Triggering *Additional Programming on VOIB 1-to-1 NAT Capable
Common Network Diagram Using Multiple Public IPs KSU Hub or Switch Router VOIB With A Static Public IP Data Network Internet Carrier Routes to Multiple Public IP Addresses Routes to Multiple Public IP Addresses Inside the Network Inside the Network
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. (RFC 3261)(RFC 3261)
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 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 2xx Success – The Action Was Successfully Received Understood Accepted 3xx Redirection – Further Action Needs To Be Taken To Complete Request 3xx Redirection – Further Action Needs To Be Taken To Complete Request 4xx Method/Client Error 4xx Method/Client Error 5xx Server Failure/Error 5xx Server Failure/Error 6xx Global 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: Invite a User or a Service to a New Session/Modify Session INVITE: Invite a User or a Service to a New Session/Modify Session ACK: Confirm Session Establishment ACK: Confirm Session Establishment OPTION: Request Information About the Capabilities of a Server OPTION: Request Information About the Capabilities of a Server BYE: End of a Session BYE: End of a Session CANCEL: Cancel a Pending Request CANCEL: Cancel a Pending Request REGISTER: Register the User Agent REGISTER: Register the User Agent
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. Call is answered and connected. Conversation takes place. Conversation takes place. We end the call and it is disconnected and We end the call and it is disconnected and the line cleared. the line cleared.
SIP Error Messages 4xx Method Failures/Client Error - Generally Authentication Failure 4xx Method Failures/Client Error - Generally Authentication Failure 401 Unauthorized - Authentication Failure 401 Unauthorized - Authentication Failure 402 Payment Required- Call Rejected 402 Payment Required- Call Rejected 403 Forbidden- Authentication Failure 403 Forbidden- Authentication Failure 5xx Server Failure/Server Errors - Server Failed to fulfill a Valid Request 5xx Server Failure/Server Errors - Server Failed to fulfill a Valid Request 503 Service Unavailable- SIP Server May Be Down 503 Service Unavailable- SIP Server May Be Down 504 Gateway Timeout- SIP Server May Be Down 504 Gateway Timeout- SIP Server May Be Down 6xx Global Failure/Global Errors – Request Cannot Be Fulfilled at Any Server 6xx Global Failure/Global Errors – Request Cannot Be Fulfilled at Any Server 600 Busy Everywhere- User Busy 600 Busy Everywhere- User Busy 606 Does Not Exist Anywhere- Unallocated Number 606 Does Not Exist Anywhere- Unallocated Number *Many 4xx Errors Report On The Station Display
SIP Call Trace Capture – Failed Call [SIP-CMD] INVITE sip: ;user=phone From: To: Contact:sip: : C>10 04, D5 B5 09 1D 3C A E E E E 0A Truncated [SIP-EVT] SIP_CALLFAIL_RESP_MSG IE_SIP_RESPONSE_CODE:403 IE_SIP_CONTACT:sip: :5060
Contact Information Training AvailableTraining Available *Vertical University (Web Based) https://university.vertical.com https://university.vertical.comhttps://university.vertical.com Software and Documents AvailableSoftware and Documents Available *VConnect *VConnect Technical SupportTechnical Support ** Online ** Phone Vertical Vertical * Requires Login * Requires Login ** Requires Tech Number