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AudioCodes’ CPE Basic Technical Training version 5.0

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AudioCodes’ CPE Basic Technical Training version 5.0

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  1. AudioCodes’ CPEBasic Technical Training version 5.0

  2. Prerequisites Basic field experience Background knowledge of PSTN & VOIP terms and concepts Good familiarity with data networking and security Exposure to voice over IP applications Basic troubleshooting methodology and experience AudioCodes CPE Basic Training

  3. Course Objectives Upon completion of this course, you will be able to perform the following tasks: • Understand the SIP/H323 protocol, methods, and how it can implemented in VoIP networks. The course is SIP focused. • Introduction to the enterprise VoIP networking (EVN) offering • Be familiar with advanced & unique ACL features • Understand SIP/H.323 supplementary and enhanced protocol features implemented by AudioCodes. • Be able to implement complex scenarios and configure AudioCodes enterprise VoIP gateway family, be able to use variety tools for troubleshooting

  4. Sources Of Information • LTRT-68804 Mediant & TP Series SIP Digital Gateways User's Manual • LTRT-65405 Media Pack SIP User's Manual • AudioCodes site: • IETF Standards

  5. Chapter 1 VoIP Brief Overview

  6. Voice Protocols Web Integration IVR Internet Protocols IP TCP RTP VoIP Services H.323 MGCP Control Protocols Speech Coding UDP RTCP Megaco SIP Network Interfaces Delay Conference Jitter Call Control Packet Loss Congestion Prioritization Network Performance VoIP General Terms

  7. PSTN IP Network Telecommuter Branch Headquarters PSTN & Data Network

  8. PSTN IP Network Telecommuter Branch Headquarters Enterprise VoIP Networking


  10. Application Protocols H.323 / SIP Stimulus Protocols MGCP based Signaling Protocols Interworking Protocols Control Protocols MGCP Media Protocols RTP / RTCP IP Telephony Protocols Signaling Gateway SS7 IP Telephony Server SCP Media Controller Media Controller IP Phone IP Client TDM TDM IP Media Gateway Media Gateway PSTN Circuit Switch PSTN Circuit Switch TDM Phone TDM Phone IP Client

  11. What does a VoIP Packet Look Like? Type of Service Total Length Ver HL Fragment Offset Identification Flag IP Header 20 octets Time to Live Protocol Header Checksum Source Address Destination Address Source Port Destination Port UDP Header 8 octets Length Checksum V P X CC M PT Sequence Number RTP Header 12 octets Time Stamp Synchronization Source ID - SSRC Voice Bits UDP = User Datagram Protocol RTP = Real-time Transport Protocol IP = Internet Protocol L2 = Layer 2 Bits

  12. Chapter 2 SIP Protocol Overview

  13. SIP - Definition What is SIP Protocol? The Session Initiation Protocol (SIP) is a signaling protocol for initiating, managing and terminating voice and video sessions across packet networks. It is Based on a client-server architecture in which clients initiate calls and servers answer calls It is an IETF RFC 3261.

  14. SDP SIP – Protocol Stack Media Performance Audio/Video Codecs SIP RTP RTCP Transport Layer/UDP

  15. Proxy Server Sip Registrar SIP Redirect Optional Entities SIP Entities User Agent/Application (both SW and HW)

  16. The endpoint entity which initiate and terminate sessions by exchanging requests and responses. The User agents consists of two components: User Agent Client UAC (originates requests) User Agent Server UAS (reply for requests) REQUEST UAS UAC RESPONSE REQUEST UAS RESPONSE UAC User Agent (UA)

  17. Receives a SIP request from a user and acts on behalf in forwarding or responding to the request. Can perform functions such as : Authentication Authorization Network Access Control Call Routing Proxy Server SIP request SIP request SIP response SIP response Media Session A Proxy interprets, and, if necessary, rewrites a request message before forwarding it. SIP Servers-Proxy

  18. Registrar Receives and accept registrations regarding current users locations. Maintains user’s whereabouts at a Location Server. Co-located with a proxy server or a redirect server May also support authentication. Redirect Server Redirects callers to other server. Maps the SIP address of the called party. Provides information about next hop to the users. Maps address to zero or more real addresses. Generates SIP responses to locate other entities. SIP Servers-cont’

  19. A Typical network Layout Example PROXY Request 1 Request 2 Response 3 User Agent User Agent PSTN IP NETWORK Response 2 SIP Overview User Agent

  20. SIP addressing Convention: user@domain. User can be: user name or phone number. Domain can be: domain name or numeric network address. Responses Responses INFO Uniform Resource Identifier URI • SIP gives you a globally reachable address. • Callees bind this address using SIP REGISTER. • Callers use it to establish real-time communication. • Examples: • sip:55551234@ • • • sip:helpdesk@

  21. Session Description Protocol What does SDP Header Specify? SDP = Session Description Protocol SDP defines Media Streams that User Agents are Capable of Receiving. Session name and purpose Time the session is active The media comprising the session Transport address/port and media format of session Session bandwidth Contact information

  22. Initiate call INVITE 1xx Confirm final response ACK 2xx Informational Release call BYE 3xx Success Cancel pending request Redirection CANCEL 4xx Failure of Request Features supported OPTIONS 5xx Server failure Register with Registrar server REGISTER 6xx Global failure From the client to the server From the server to the client SIP Messages Requests (Methods) Responses

  23. receive RTP G.711-encoded audio at SIP Message Structure Requests Methods Response Status Responses INVITE SIP/2.0 Via: SIP/2.0/UDP From: Ronen <> To: Trainee <> Call-ID: CSeq: 1 INVITE Subject: SIP Training Contact: Ronen <>; Content-Type: application/sdp Content-Length: 147 SIP/2.0 200 OK Via: SIP/2.0/UDP From: Ronen <>;tag=76341 To: trainee <> Call-ID: CSeq: 1 INVITE Subject: SIP Training Contact: Ronen <>; Content-Type: application/sdp Content-Length: 134 2xx 3xx 4xx v=0 o=ACL 2890844526 2890844526 IN IP4 s=Phone Call c=IN IP4 t=0 0 m=audio 6010 RTP/AVP 0 a=rtpmap:0 PCMU/8000 v=0 o=ACL 2890844526 2890844527 IN IP4 s=Phone Call c=IN IP4 t=0 0 m=audio 6030 RTP/AVP 0 a=rtpmap:0 PCMU/8000 5xx 6xx

  24. SIP Basic Methods Message that Indicates that the user or service is being invited to participate in a session. Responses INVITE The ACK request confirms that the client has received a final response to an INVITE request ACK BYE The client sends this message to release the Call. 2xx Cancels a request in progress; has no effect on an established call. CANCEL 3xx Query the capabilities of a call agent. OPTIONS Client uses the register method to register its address with the SIP server REGISTER 4xx

  25. General header fields Responses The call-id uniquely identifies a particular invitation of a particular client. Typically uses a 32-bit cryptographically random numbers CALL-ID Composed of an unsigned sequence number and the method name. Incremented at each new request Starts at a random value. Cseq 2xx • Contains the name and the address of the originator • of the request From Contains the name and the address of the called party. 3xx TO Via Shows the route taken by request. 4xx Content Type Provides information about media type of message body Session Expires • conveys the session interval for a SIP session

  26. SIP Extension Methods Responses Mid-call signaling (RFC 2976) Used for DTMF, hook flash , special usage for IPM2000. INFO Call transfer (RFC 3515) REFER SUBSCRIBE/ NOTIFY Presence and other services (RFC 3265) 2xx Provisional Reliable Responses ACKnowledgment (RFC 3262) PRACK 3xx Update parameters of a session (accept only) UPDATE 4xx

  27. SIP Responses Informational Redirection Server Failure 300 Multiple Choices 301 Moved Perm 302 Moved Temp 380 Alternative Serv 100 Trying 180 Ringing 181 Call forwarded 182 Queued 183 Session Progress (Early Media) 504 Timeout 503 Unavailable 501 Not Implemented 500 Server Error Request Failure Global Failure 600 Busy Everywhere 603 Decline 604 Doesn’t Exist 606 Not Acceptable 200 OK 202 Accepted Success • 400 Bad Request • 401 Unauthorized • 403 Forbidden • 404 Not Found • 405 Bad Method • 415 Unsupp Content • 420 Bad Extensions • 486 Busy Here

  28. INVITE: From: CallID: [unique identity of call] Cseq: [command sequence number] Session Description Protocol V = [protocol version] O =[creator, session ID] T = [time session is active] C = [Connection IP Info] M =[media type 1 and transport address] M =[media type 2 and transport address] INVITE + SDP 180 Ringing 100 Trying 200 OK + SDP ACK Basic Call Flow 200 100

  29. Chapter 3 Specific Concepts AudioCodes Gateways

  30. TEL x IP Sides • TEL Side is everything related to the old telephony interfaces: FXS, FXO, E1 (PRI, Q.SIG, R2 etc.) • IP Side is the gateway function: packetization, codecs, communication with the proxy etc. • A gateway always has both sides TEL Side IP Side FXS Interfaces Gateway FXO Interfaces IP Network E1 Interfaces

  31. Endpoints • Endpoints are the basic TEL interfaces • Each FXS port is an endpoint • Each FXO port is an endpoint • Each timeslot is an endpoint • It’s mandatory to assign a phone number to any endpoint that will be used • No phone number, no call Gateway FXS Endpoint #1 FXS Endpoint #2 FXS Endpoint #3 FXO Endpoint #4 FXO Endpoint #5 IP Network FXO Endpoint #6 Endpoint #7 E1 Interface Endpoint #N

  32. User Gateways x Trunk Gateways • User Gateways dedicate one endpoint (and only one) to each human user • Trunk Gateways dedicate one (or more) endpoint to a group of human users • These concepts relate mostly to the TEL side (mostly, not only…) • Typically, FXS gateways are configured as User Gateways • Endpoint Phone Numbers are the users’ addresses (they are routed in the NGN) • Typically, FXO and E1 gateways are configured as Trunk Gateways • Endpoint Phone Numbers are typically used for endpoint identification only – not as addresses. TEL #1 User #1 User Gateway TEL #2 User #2 IP Network TEL #3 User #3 TEL #1 Trunk Gateway PBX, Switch... IP Network TEL #2

  33. Trunk Groups (or Hunt Groups) • Applicable only to Trunk Gateways • Trunk Gateways dedicate one (or more) endpoint to a group of human users • These endpoints are grouped in Trunk Groups • Trunk Groups may be also called Hunt Groups Trunk Group (or Hunt Group) Human Users TEL #1 PBX, Switch... Trunk Gateway ... IP Network TEL #M

  34. Chapter 4 MPFXS11X

  35. Chapter Objectives Upon completion of this chapter, you will be able to perform the following tasks: • Basic configuration of the MP-FXS. • Understand the SIP call flow and its basic implementations. • Use BootP & TFTP Servers. • Troubleshoot using the Syslog Server. • Understand Proxy Function’s.

  36. Place Call Configuration Process • Required Steps to Install the MP-11x: MP Connectivity Changing IP address MP Configuration

  37. MP Connectivity Front Panel Channels Status Leds Lan Led FAIL Ready On Off hooked Blinking Ringing Fast Blinking Line malfunction On Valid 10/100 connection Off No uplink On Failure Off working On Device powered Off System failure

  38. Power Supply Socket RS-232 Port Reset Button 10/100 Base-T RJ-45Port 8 RJ-11 FXS Ports MP Connectivity Rear Panel

  39. MP-11x IP Setting • Changing PC’s IP address and Subnet Mask • to correspond with the MP-11x’s factory default • IP address and Subnet Mask. • DefaultMP11x IP factory • IP Address: FXS • FXO • Subnet Mask : Both255.255.0.0 IP_SETTING

  40. Assigning IP Address Assigning an IP address to the MP-1xx: • HTTP using switch / hub or using direct PC connection. • BootP Application

  41. Assigning IP Address • Serial via the RS-232 port SCP IP [ip_address] [subnet_mask] [default_gateway] &SAR • DHCP DHCPEnable = 1 http://acl_<serial number> • Voice Menu Guidance : Lift the handset and dial ***12345

  42. MP Basic Configuration • Use the quick setup menu • Configure endpoints telephone numbers • Match maximum digit number • Enter calls routing using Tel to IP routing Table MP_CONFIGURATION

  43. BootP Application BootP Definition: The Bootstrap Protocol allows a host to configure itself dynamically. This protocol provides 3 services: • IP address assignment. • Detection of the IP address for a serving machine. • A file to be loaded and executed by the client machine. • Relevant parameters: • AudioCodes' BootP/TFTP Application. • Device Mac address. • Files.

  44. Configuring BootP & TFTP Server TFTP Server Enabling the Server BootP Server IP Address Home Directory Change Retransmissions to 20!

  45. Configuring BootP & TFTP Server Client Configuration MAC Address Enable Device IP Settings TFTP IP Image file (*.cmp) Use Flash Burn (“ –fb”) INI File

  46. Configuring BootP & TFTP Server

  47. Troubleshooting BootP/TFTP Take the following steps to resolve BootP/TFTP issues: • Validate BootP/TFTP server and client configuration. • Validate Ethernet cable and that the BootP server is configured on the same subnet. • Collect an Ethereal/WireShark trace and validate BootP/TFTP process. • Use a crossover Ethernet cable to eliminate any network issues. • Check the Max Retransmissions (Edit->Preferences->TFTP Server)

  48. Configurations Options • Embedded Web Server • A configuration file referred to as the ini file • AudioCodes’ Element Management System (EMS) • An SNMP-based program

  49. Embedded Web Server Buttons: Submit / Reset /Save Configuration ! (Not changeable on-the-fly and require reset) HTTPS:

  50. Restoring Default Parameters To restore networking parameters to their initial state take these 5 steps: 1. Press in the ‘Reset’ button uninterruptedly for more than 6 seconds; 2. The gateway is restored to its factory settings (username: ‘Admin’, password: ‘Admin’) 3. Assign the MP-11x IP address. 4. Load your previously ini file 5. Press again on the ‘Reset’ button (short period)