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 3 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 4 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. 5 Direct Internet Access Diagram Data Network Carrier Internet Router Hub or Switch KSU VOIB With A Static Public IP 6 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 15 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 17