VoIP VoIP /05/29

Transcription

1 1 VoIP /05/29

2 2 0.9 M-Taiwan

3 VOIP RFC SIP CONFORMANCE TEST TTCN MOS PESQ VOIP PROXY SERVER REDIRECT SERVER REGISTRAR USER AGENT VOIP VOIP VOIP NETWORK ARCHITECTURE AND ELEMENTS VOIP SERVICE INTEROPERABILITY ENVIRONMENT TEST CASE OVERVIEW DETAILED TEST CASE DESCRIPTION

4 4 SIP...11 VOIP...16 VOIP ( )...67 ( )...68

5 M-Taiwan All IP Network IP Voice over IP Video over IP Multimedia Messaging IP Phone SIP Proxy IP-PBX Media Server Softswitch VoIP Voice over IP SIP Session Initiation Protocol VoIP VoWLAN Voice over Wireless LAN VoIP VoIP VoIP

6 6 2 VoIP 2.1 RFC VoIP SIP Session Initiation Protocol RFC 3261 Session Initiation Protocol(SIP) RFC Reliability of Provisional Responses in the SIP RFC Session Initiation Protocol - Locating SIP Servers RFC An Offer/Answer Model with the Session Protocol (SDP) RFC Session Initiation Protocol Specific Event Notification RFC Session Initiation Protocol Basic Call Flow Examples RFC SIP Public Switched Telephone Network-PSTN Call Flows RFC E.164 number and DNS RFC ENUM Service Registration for SIP RFC The SIP INFO Method RFC The SIP UPDATE method RFC The Session Initiation Protocol Refer Method RFC The Early Session Disposition Type of SIP SIP (RFC3665) VoIP Successful New Registration Update of Contact List Request for Current Contact List Cancellation of Registration

7 Unsuccessful Registration Successful Session Establishment Session Establishment Through Two Proxies Session with Multiple Proxy Authentication Successful Session with Proxy Failure Session Through a SIP ALG Session via Redirect and Proxy Servers with SDP in ACK Session with re-invite (IP Address Change) Unsuccessful No Answer Unsuccessful Busy Unsuccessful No Response from User Agent Unsuccessful Temporarily Unavailable SIP Conformance Test SIP VoIP ETSI TIPHON SIP conformance test VoIP SIP RFC 3261 Test Purpose ETSI RTS/TIPHONE SIP RFC 3261 Test Cases ETSI RTS/TIPHONE (TTCN-3)

8 8 TIPHON SIP conformance status Status Mandatory Recommended Optional Recommended/Optional Mandatory ( User Agent Server) Recommended 2.3 TTCN-3 TTCN-3 Testing and Test Control Notation version 3 ETSI TTCN-3 TTCN-3 ITU-T Z.140 ETSI ES TTCN-3 TTCN MOS, 98% VoIP 63% PSTN VoIP 1996 ITU-T MOS Mean Opinion Score MOS 16

9 9 MOS MOS 2.5 PESQ 2001 ITU-T P.862 PESQ Perceptual Evaluation of Speech Quality PESQ / Codec PESQ MOS Telecommunications Industry Association MOS 4.0 VoIP PESQ

10 10 3 VoIP SIP VoIP VoIP SIP Server Proxy Registrar Redirect Server RFC 3261 Session Initiation Protocol(SIP) RFC Reliability of Provisional Responses in the SIP RFC Session Initiation Protocol - Locating SIP Servers RFC An Offer/Answer Model with the Session Protocol (SDP) RFC Session Initiation Protocol Specific Event Notification RFC Session Initiation Protocol Basic Call Flow Examples RFC SIP Public Switched Telephone Network-PSTN Call Flows RFC E.164 number and DNS RFC ENUM Service Registration for SIP RFC The SIP INFO Method RFC The SIP UPDATE method RFC The Session Initiation Protocol Refer Method RFC The Early Session Disposition Type of SIP IETF SIP SIP Caller SIP UAC - User Agent Client Callee SIP UAS - User Agent Server SIP Request Message SIP SIP Server SIP Response Message

11 11 Media Session SIP SIP SIP Proxy Server Registrar Redirect Server SIP Proxy Server 1 Proxy Server 4 Proxy Server 5 Redirect Server 2 Location Server 3 SIP SIP Location Server SIP SIP SIP Request SIP Response RTP Media Stream User Agent Client (Caller) Proxy Server1 Redirect Server2 Location Server3 Proxy Server4 Proxy Server5 User Agent Client (Caller) User Agent Server (Callee) User Agent Client : UAC SIP Client

12 12 Server User Agent Server UAS User Agent : UA 3.1 Proxy Server SIP Proxy Server VoIP Compliant with RFC Backward compatible with RFC Support SIP User Agents (e.g., IP Softphone, IP Hard Phone, IP Videophone) to call establish, modify and terminate. Support Record-Route routing TCP and UDP conversion. Strict and loose routing compatible. DNS SRV record query. Loop detection and spiral. (For RFC 2543 is mandatory, for RFC 3261 is option.) Support both stateful and stateless modes. Sequential and parallel forking. Support parallel CANCEL function. Support authentication and authorization mechanism. 3.2 Redirect Server SIP Redirect Server VoIP Support reply response message with 3xx status code. Support route rule definition.

13 Registrar SIP Registrar Server VoIP proxy redirect server Support User Agent register and un-register functions. Support 3rd party registration function. Support authentication mechanism. 3.4 User Agent User Agent Softphone IP Phone User Agent RFC 3261 Signaling features: - End-to-end call establishment, maintaining and terminating as per RFC 3261(SIP). Backward compatible with RFC Dynamic negotiation of session parameters using RFC 2327(SDP). - Support RFC 3264 (An Offer/Answer Model with the Session Protocol). - Support all RFC 2543/3261 SIP headers parsing. - Support HTTP Digest (MD5 Hash as described in RFC 2617) authentication mechanism. - Support User Agent register/un-register/registration query functions. - Support User Agent Client (UAC) make/cancel call, and User Agent Server (UAS) answer/reject call functions. Both UAC and UAS can terminate calls. - Support DNS SRV record query, using DNS procedures to allow a client to resolve a SIP Uniform Resource Identifier (URI) into the IP address, port, and transport protocol of the next hop to contact, as per RFC Media processing features:

14 14 - Real-time audio/video transfer as per RFC 1889(RTP/RTCP). - Compliant to RFC 1890(RTP Profile for Audio and Video Conferences with Minimal Control). Configurable features: - Registrar server - Outbound Proxy server - Codec selection. - System microphone and speaker volume control. Support peer-to-peer call connection and proxy call routing. Support both TCP and UDP transport.

15 15 4 VoIP VoIP IPv4 SIP SDP RFC 3666 (SIP Public Switched Telephone Network-PSTN Call Flows) VoIP 1. e.164 e.164 ( e SIP 3. VoIP VoIP SIP SIP Private Network Firewall SIP 4.

16 VoIP Intranet Private IPv4 Domain SIP Servers (A) VoIP A SIP UA (A) NAT/Firewall Internet Public IPv4 Domain SIP Servers (B) VoIP B SIP UA (B) VoIP

17 VoIP VoIP Center / Relay/Proxy server 4E1 Recording server 2E1 Billing server IP-PBX E1/12 FXO IP-PBX E1/4 FXO/8 FXS

18 Register IP-PBX VoIP Center Proxy IP-PBX Register request Proxy Proxy message code 401 IP-PBX IP-PBX Register request with password Proxy Proxy 200 OK IP-PBX CPE IP-PBX CPE Register request IP-PBX IP-PBX message code 401 CPE CPE Register request with password IP-PBX IP-PBX 200 OK CPE IP-PBXA IP-PBXB Call setup message If successful IP-PBXA INVITE Proxy IP-PBXB IP-PBXB 200 OK Proxy IP-PBXA If not successful IP-PBXA INVITE Proxy Message NOT found 404 IP-PBXA RTP packet: IP-PBXA to IP-PBXB

19 19 IP PBXA INVITE SIP Proxy IP PBXB 100 Trying F2 INVITE Trying F4 183 Progress (Ring back) Both Way RTP Media 200 OK F11 ACK Both Way RTP Media BYE 200 OK F Progress (Ring) Early Media 200 OK F10 ACK BYE 200 OK F16 VoIP

20 VoIP 1. SIP Conformance Certification SIP 2. Voice Quality Certification IP Phone, Gateway

21 * 8 # * 8 # 5 VoIP 21 VoIP VoIP VoIP Service Provider A VoIP Service Provider B Soft Phone Media Server Call Server Call Server Soft Phone Media Server IP Phone Application Server PSTN GW PSTN X Extension Enterprise PBX Application Server IP Phone PSTN GW IP IP PSTN PSTN IP IP PSTN CDR Home 1. VoIP VoIP

22 Network Architecture and Elements SIP-Terminal Terminal B User B Firewall/NAT Media Gateway Controller (MGC) SIP-Terminal SIP-Proxy IP SIP-Relay-Server SIP-Access-Gateway SUT User A PSTN-Gateway Registrar Analog Terminal C5 PSTN STP PSTN-Terminal

23 VoIP Service Interoperability Environment SIP-Terminals / Access-Gateways System (Terminal / Gateway) Make Software Hardware Other Network Elements Network Element SIP-Proxy Registrar SIP-Relay-Server Make ICL/ITRI ICL/ITRI ICL/ITRI Software OS Application Hardware Network Element MGC PSTN-Gateway Make ICL/ITRI Software OS Application Hardware

24 Test Case Overview

25 25

26 26

27 27

28 5.4 Detailed Test Case 28 This clause contains the detailed description of the test cases listed in the table on the previous page.

29 29 SIP_IOP_RG_SR_001 SUT registers at registrar. To verify that SUT is able to register at a registrar. 1. Test setup according to figure SUT is authorized to use registrar and proxy services. 1 User A powers up SUT. V A R I A T I O N S none 1 SUT registers at registrar.

30 30 SIP_IOP_RG_SR_002 SUT registers again at registrar before its registration becomes invalid; the SUT suggests an expiration interval either in the Expires header field or an expires Contact header field. To verify that SUT sends a new REGISTER request before its registration becomes invalid. 1. Test setup according to figure SUT is authorized to use registrar and proxy services. 3. Expiration interval in SUT set to T. 1 User A powers up SUT. V A R I A T I O N S none 1 SUT registers at registrar with a REGISTER request containing an expiration interval of T (in the Expires header field or an expires Contact header field). 1 Registrar reads T and updates expires interval for SUT. 1 SUT sends a new REGISTER request less than T seconds after the first REGISTER request was send out.

31 31 SIP_IOP_RG_SR_003 SUT does NOT suggest any expiration interval in its REGISTER request and accepts the default value from the registrar. To verify that SUT accepts a default expiration interval from the registrar. 1. Test setup according to figure SUT is authorized to use registrar and proxy services. 3. SUT is in a mode NOT to suggest an expiration interval in the Expires header field or an expires Contact header field. 4. Default expiration interval in registrar set to TD (parameter in ICL proxy: DEFAULT_REGISTER_EXP). 1 User A powers up SUT. V A R I A T I O N S none 1 SUT registers at registrar with a REGISTER request than does NOT contain an expiration interval in the Expires header field or an expires Contact header field. 1 Registrar returns the default value TD for the expiration interval in the 200 OK. 1 SUT sends a new REGISTER request less than TD seconds after the first REGISTER request was send out.

32 32 SIP_IOP_RG_UR_001 SUT is NOT authorized to register at the registrar. To verify behavior of SUT on receipt of an UNAUTHORIZED (401). 1. Test setup according to figure SUT is NOT authorized to use registrar. 1 User A powers up SUT. V A R I A T I O N S none 1 SUT sends a REGISTER request and the registrar responds with UNAUTHORIED (401). 1 SUT should re-originate the REGISTER request with the proper credentials (included in the Authorization header field).

33 33 SIP_IOP_RG_UR_002 SUT is NOT authorized to use the proxy services. To verify behavior of SUT on receipt of a PROXY AUTHENTICATION REQUIRED (407). 1. Test setup according to figure SUT is authorized to use registrar but NOT the proxy services. 1 User A powers up SUT. V A R I A T I O N S none 1 SUT sends a REGISTER request and the proxy responds with ROXY AUTHENTICATION REQUIRED (407). 1 SUT should re-originate the REGISTER request with the proper credentials (included in the Authorization header field).

34 34 SIP_IOP_RG_UR_003 To verify behavior of SUT on receipt of INTERVAL TOO BRIEF (423). 1. Test setup according to figure SUT is authorized to use registrar and proxy services. 3. Expiration interval in SUT < minimum expiration interval in registrar (parameter in ICL proxy: MIN_REGISTER_EXP) 1 User A powers up SUT. V A R I A T I O N S none 1 SUT registers at registrar. 1 Registrar sends INTERVAL TOO BRIEF (423). 1 SUT may send a new REGISTER request with an expiration interval > minimum expiration interval in registrar.

35 35 SIP_IOP_RG_DR_001 SUT removes its registration. To verify that a call can be established successfully from the SUT to Terminal B; Record Route in proxy = enabled. 1. Test setup according to figure Users A and B registered. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI {Modifier 2}. 3 User B accepts call and goes off-hook. 4 Users A releases call according to test case SIP_IOP_BC_CR_002. V A R I A T I O N S none 1 SUT registers at registrar. 2 SUT sends REGISTER request with expiration interval set to 0.

36 36 SIP_IOP_BC_SC_001 User A (using the SUT) calls User B (Terminal B). Proxy Authentication = required {Modifier 1}; Record Route = enabled. To verify that a call can be established successfully from the SUT to Terminal B; Record Route in proxy = disabled. 3. Test setup according to figure Users A and B registered. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI {Modifier 2}. 3 User B accepts call and goes off-hook. 4 User A releases call according to test case SIP_IOP_BC_CR_002. V A R I A T I O N S Modifier 1 Modifier 2 Terminals of at least 3 different vendors. SIPS URI 1 User A hears dial tone. 2 Terminal B is ringing. 3 SUT sends ACK to proxy (Record Route = enabled!). 3 Speech path established.

37 37 SIP_IOP_BC_SC_002 User A (using the SUT) calls User B (Terminal B) {Modifier 1}. Proxy Authentication = required; Record Route = disabled. To verify that a call can be established successfully from the SUT to Terminal B; Record Route in proxy = disabled. 5. Test setup according to figure Users A and B registered. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI {Modifier 2}. 3 User B accepts call and goes off-hook. 4 User A releases call according to test case SIP_IOP_BC_CR_002. V A R I A T I O N S Modifier 1 Modifier 2 Terminals of at least 3 different vendors. SIPS URI 1 User A hears dial tone. 2 Terminal B is ringing. 3 SUT sends ACK to Terminal B (Record Route = disabled!). 3 Speech path established.

38 38 SIP_IOP_BC_SC_003 User B (using Terminal B) calls User A (SUT). Proxy Authentication = required; Record Route = enabled. To verify that a call can be established successfully from Terminal B to the SUT; Record Route in proxy = enabled. 1. Test setup according to figure Users A and B registered. 1 User B goes off-hook. 2 User B dials number of User B and sends dialed number using a SIP URI {Modifier 1}. 3 User A accepts call and goes off-hook. 4 User B releases call according to test case SIP_IOP_BC_CR_003. V A R I A T I O N S Modifier 1 SIPS URI 1 User B hears dial tone. 2 SUT is ringing. 3 Terminal B sends ACK to proxy (Record Route = enabled!). 3 Speech path established.

39 39 SIP_IOP_BC_SC_004 User B (using Terminal B) {Modifier 1} calls User A (SUT). Proxy Authentication = required; Record Route = disabled. To verify that a call can be established successfully from Terminal B to the SUT; Record Route in proxy = disabled. 1. Test setup according to figure Users A and B registered. 1 User B goes off-hook. 2 User B dials number of User B and sends dialed number using a SIP URI {Modifier 2}. 3 User A accepts call and goes off-hook. 4 User B releases call according to test case SIP_IOP_BC_CR_004. V A R I A T I O N S Modifier 1 Modifier 2 Terminals of at least 3 different vendors. SIPS URI. 1 User B hears dial tone. 2 SUT is ringing. 3 Terminal B sends ACK to SUT (Record Route =disabled!). 3 Speech path established.

40 40 SIP_IOP_BC_SC_005 User A (using the SUT) calls User B (PSTN-Terminal). Proxy Authentication = required; Record Route = disabled. To verify that a call can be established successfully from the SUT to a PSTN-Terminal. 1. Test setup according to figure Users A (SUT) registered. 1 User A goes off-hook. 2 User A dials number of User B (PSTN-Terminal) and sends dialed number using a SIP URI {Modifier 1}. 3 User B accepts call and goes off-hook. 4 User A releases call according to test case SIP_IOP_BC_CR_001. V A R I A T I O N S Modifier 1 SIPS URI 1 User A hears dial tone. 2 Terminal B is ringing. 3 SUT sends ACK to PSTN-Gateway (or MGC?) (Record Route = disabled!). 3 Speech path established.

41 41 SIP_IOP_BC_SC_006 User A (PSTN-Terminal) calls User B (SUT). Proxy Authentication = required; Record Route = disabled To verify that a call can be established successfully from a PSTN-Terminal to the SUT. 1. Test setup according to figure Users B registered. 1 User A (PSTN-Terminal) goes off-hook. 2 User A dials number of User B and sends dialed number. 3 User B accepts call and goes off-hook. 4 User A releases call according to test case SIP_IOP_BC_CR_003. V A R I A T I O N S none 1 User A hears dial tone. 2 Terminal B is ringing. 3 PSTN-Gateway sends ACK to SUT (or MGC?) (Record Route = disabled!). 3 Speech path established.

42 42 SIP_IOP_BC_UC_001 User A (using the SUT) calls User B (Terminal B). Terminal B is busy. To verify behavior of SUT (= caller) when callee busy. 1. Test setup according to figure Users A and B registered. 3. Terminal B is busy. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI. 3 User A releases call. V A R I A T I O N S none 1 User A hears dial tone. 2 Terminal B sends BUSY HERE (486). 2 User A hears busy tone.

43 43 SIP_IOP_BC_UC_002 User A (using the SUT) calls User B (Terminal B). User B does not answer. To verify behavior of SUT (= caller) when callee does not answer. 1. Test setup according to figure Users A and B registered. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI. 3 User B doesn t accept call. 3 User A releases call. V A R I A T I O N S none 1 User A hears dial tone. 2 Terminal B is ringing. 3 Terminal B sends REQUEST TIMEOUT (408).

44 44 SIP_IOP_BC_UC_003 User A (using the SUT) calls User B (Terminal B). Terminal B switched off for a period > expiration time. To verify behavior of SUT (= caller) when callee is switched off. 1. Test setup according to figure Users A and B registered. 3. Terminal B switched off for a period > expiration time. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI. 3 User A releases call. V A R I A T I O N S none 1 User A hears dial tone. 2 Proxy sends NOT FOUND (404).

45 45 SIP_IOP_BC_UC_004 User A (using the SUT) calls User B (Terminal B). User B declines to accept call. To verify behavior of SUT (= caller) when callee declines to accept call. 1. Test setup according to figure Users A and B registered. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI. 3 User B explicitly declines to accept the call. 4 User A releases call. V A R I A T I O N S none 1 User A hears dial tone. 2 Terminal B is ringing. 3 Terminal B sends DECLINE (603).

46 46 SIP_IOP_BC_UC_005 User A (using the SUT) calls User B (Terminal B). Terminal B replies with UNSUPPORTED MEDIA TYPE (415). To verify behavior of SUT on receipt of UNSUPPORTED MEDIA TYPE (415). 1. Test setup according to figure Users A and B registered. 1 User A goes off-hook. 2 User A dials number of User B and sends dialed number using a SIP URI. INVITE request contains a encoding method which is not being supported by Terminal B. 3 User B accepts call and goes off-hook. 4 User A releases call. V A R I A T I O N S none 1 User A hears dial tone. 2 Terminal B is ringing. 3 Terminal B sends UNSUPPORTED MEDIA TYPE (415). 3 SUT should retry sending the INVITE request, this time only using encodings listed in the Accept-encoding header field in the Speech path established

47 47 SIP_IOP_BC_CR_001 User A (using the SUT) terminates established call; Record Route = enabled. To verify that a call can be successfully terminated from the SUT (= caller); Record Route in proxy = enabled. 1. Test setup according to figure Users A and B registered. 3. Call established according to test case SIP_IOP_BC_SC_001 {Modifier 1}. 1 User A goes on-hook. V A R I A T I O N S Modifier 1 SIPS URI 1 SUT sends BYE request to proxy (Record Route = enabled!). 1 All resources are cleared.

48 48 SIP_IOP_BC_CR_002 User A (using the SUT) terminates established call {Modifier 1}; Record Route = disabled. To verify that a call can be successfully terminated from the SUT (= caller); Record Route in proxy = disabled. 1. Test setup according to figure Users A and B registered. 3. Call established according to test case SIP_IOP_BC_SC_002 {Modifier 2} 1 User A goes on-hook. V A R I A T I O N S Modifier 1 Modifier 2 Terminal on terminating side (Terminal B) from at least 3 different vendors. SIPS URI 1 SUT sends BYE to Terminal B (Record Route = disabled!). 1 All resources are cleared.

49 49 SIP_IOP_BC_CR_003 User B (using Terminal B) terminates established call; Record Route = enabled. To verify that a call can be successfully terminated from Terminal B (= caller); Record Route in proxy = enabled. 1. Test setup according to figure Users A and B registered. 3. Call established according to test case SIP_IOP_BC_SC_003 {Modifier 1}. 1 User B goes on-hook. V A R I A T I O N S Modifier 1 SIPS URI 1 Terminal B sends BYE to proxy (Record Route = enabled!). 1 All resources are cleared.

50 50 SIP_IOP_BC_CR_004 User B (using Terminal B) {Modifier 1} terminates established call; Record Route = disabled To verify that a call can be successfully terminated from Terminal B (= caller); Record Route in proxy = disabled 1. Test setup according to figure Users A and B registered. 3. Call established according to test case SIP_IOP_BC_SC_004 {Modifier 2}. 1 User B goes on-hook. V A R I A T I O N S Modifier 1 Modifier 2 Terminal on terminating side (Terminal B) from at least 3 different vendors. SIPS URI 1 Terminal B sends BYE to SUT (Record Route = disabled!). 1 All resources are cleared.

51 51 SIP_IOP_BC_CR_005 Users A and B terminate the call at the same time; Record Route = disabled. To verify call termination when caller und callee hang up simultaneously. 1. Test setup according to figure Users A and B registered. 3. Call established. 1 User A and User B go on-hook exactly at the same time. V A R I A T I O N S none 1 SUT and Terminal B send BYE to each other (Record Route = disabled!). 1 All resources are cleared.

52 52 SIP_IOP_BC_CR_006 User A (using the SUT) terminates call attempt before Terminal B answers; Record Route = disabled. To verify that call establishment can be successfully cancelled from the SUT. 1. Test setup according to figure Users A and B registered. 1 User A goes off-hook. 2 User A dials number of User B, sends dialed number using a SIP URI and immediately goes on-hook again. Ensure that Terminal B did NOT send a final response (200 OK) yet. V A R I A T I O N S none 1 User A hears dial tone. 2 SUT sends CANCEL to proxy. 2 All resources are cleared.

53 53 SIP_IOP_BC_CR_007 User B (using the Terminal B) terminates call attempt before SUT answers; Record Route = disabled. To verify that call establishment can be successfully cancelled from Terminal B. 1. Test setup according to figure Users A and B registered. 1 User B goes off-hook. 2 User B dials number of User A, sends dialed number using a SIP URI and immediately goes on-hook again. Ensure that Terminal A did NOT send a final response (200 OK) yet. V A R I A T I O N S none 1 User B hears dial tone. 2 Terminal B sends CANCEL to proxy. 2 All resources are cleared.

54 54 SIP_IOP_AC_RD_001 User A (using the SUT) calls User B (Terminal B) and gets redirected to Terminal C. Proxy Authentication = required; Record Route = disabled To verify behavior of SUT on receipt of 3xx. 1. Test setup according to figure Users A, B and C registered. 3. Redirection in proxy = enabled. 1 User A goes off-hook. 2 User A dials number of User B, sends dialed number using a SIP URI and gets redirected to Terminal C. 3 User B at Terminal C accepts call and goes off-hook. 4 User A releases call. V A R I A T I O N S Modifier 1 Modifier 2 Proxy Authentication = NOT required SIPS URI 1 User A hears dial tone. 2 Proxy sends 3xx to SUT. 2 SUT sends new INVITE request to Terminal C. 3 Speech path established.

55 55 SIP_IOP_SM_SM_001 User A (using the SUT, caller) in a call with User B, changes the encoding method {Modifier 1}. Proxy Authentication = required; Record Route = disabled. To verify that a session modification can be performed successfully, initiated by SUT (= caller). 1. Test setup according to figure Users A and B registered. 3. Call established between Users A (= caller) and B. 4. Record Route in proxy = disabled. 1 User A changes encoding method for existing call. V A R I A T I O N S Modifier 1 IP address change. 1 SUT sends re-invite to Terminal B (Record Route = disabled!). 1 Encoding method for established call changed.

56 56 SIP_IOP_SM_SM_002 User A (using the SUT) in a call with User B (= caller), receives a re-invite from User B to change the encoding method {Modifier 1}. Authentication = required; Record Route = disabled To verify that a session modification can be performed successfully, initiated by Terminal B (= caller). 1. Test setup according to figure Users A and B registered. 3. Call established between Users A and B (= caller). 4. Record Route in proxy = disabled. 1 User B (Terminal B) changes encoding method for existing call. V A R I A T I O N S Modifier 1 IP address change 1 Terminal B sends re-invite to SUT (Record Route = disabled!). 1 Encoding method for established call changed.

57 57 SIP_IOP_SM_UM_001 User A (using the SUT, caller) in a call with User B (Terminal B) changes the encoding method. However the requested encoding method is not supported by Terminal B which therefore rejects the new session and responds with NOT ACCEPTABLE HERE (488). Proxy Authentication = required; Record Route = disabled. To verify behavior of SUT on receipt of NOT ACCPTABLE HERE (488); SM initiated by SUT. 1. Test setup according to figure Users A and B registered. 3. Call established between Users A and B. 4. Record Route in proxy = disabled. 1 User A changes encoding for existing call. New encoding method is not supported by Terminal B. V A R I A T I O N S none 1 SUT sends re-invite to Terminal B (Record Route = disabled!). 1 Terminal B rejects new encoding method and sends NOT ACCEPTABLE HERE (488).

58 58 SIP_IOP_SM_UM_002 User A (using the SUT, caller) is in a call with User B (Terminal B). User B changes the encoding method. However the requested encoding method is not supported by SUT which therefore rejects the new session and responds with NOT ACCEPTABLE HERE (488). Proxy Authentication = required; Record Route = disabled. To verify behavior of SUT on receipt of a re-invite with a not supported encoding method. 1. Test setup according to figure Users A and B registered. 3. Call established between Users A and B. 4. Record Route in proxy = disabled. 1 User A changes encoding for existing call. V A R I A T I O N S none 1 Terminal B sends re-invite to SUT (Record Route = disabled!). 1 SUT rejects new encoding method and sends NOT ACCEPTABLE HERE (488).

59 59 SIP_IOP_QC_SQ_001 User A (using the SUT, caller) queries Terminal B as to its capabilities. Proxy Authentication = required. To verify that SUT can query another Terminal for its capabilities. 1. Test setup according to figure Users A and B registered. 1 User A (SUT) queries for capabilities of Terminal B. V A R I A T I O N S none 1 SUT sends OPTIONS request to Terminal B. 1 Terminal B responds with 200 OK containing Allow, Accept, Accept- Encoding etc. header fields listing the supported options.

60 60 SIP_IOP_QC_SQ_002 User A (using the SUT, caller) queries proxy as to its capabilities. Proxy Authentication = required. To verify that SUT can query a proxy for its capabilities. 3. Test setup according to figure User A registered. 1 User A (SUT) queries for capabilities of a proxy. V A R I A T I O N S none 1 SUT sends OPTIONS request to proxy. 1 Proxy responds with 200 OK containing Accept, Accept-Encoding etc. header fields listing the supported options.

61 61 SIP_IOP_QC_SQ_003 User A (using the SUT, caller) receives an OPTIONS request to query its capabilities. Proxy Authentication = required; Record Route = disabled To verify that SUT can respond to a query for capabilities from another Terminal. 1. Test setup according to figure Users A and B registered. 1 User B (Terminal B) queries for capabilities of SUT. V A R I A T I O N S none 1 Terminal B sends OPTIONS request to SUT. 1 SUT responds with 200 OK containing Allow, Accept, Accept-Encoding etc. header fields listing the supported options.

62 62 SIP_IOP_SS_CH_001 Call Hold User B (using the SUT, callee) places the established call on hold. Proxy Authentication = required; Record Route = enabled To verify that a call establishment can be successfully held from SUT. 1. Test setup according to figure Users A and B registered. 1 User A establishes a session with User B (using the SUT). 2 User B places the call on hold. 3 After a while, User B takes the call off hold. 4 User A terminates the session with User B. V A R I A T I O N S None 1 Speech path established. 2 SUT sends a RE-INVITE with a new SDP offer that includes the attribute sendonly. 2 User A accepts the Hold request with an SDP answer that includes the attribute recvonly 3 SUT takes the call off Hold by sending another RE_INVITE with a new SDP offer that does not include anymore the attribute sendonly. 3 User A accepts the request with an SDP answer that does not include anymore the attribute recvonly. 4 User A sends BYE request to proxy. 4 All resources are cleared.

63 63 SIP_IOP_SS_TR_001 Call Transfer Unattended User B calls User A (using the SUT). User A performs an unattended transfer of User B to User C. Proxy Authentication = required; Record Route = enabled To verify that a call establishment can be successfully transferred from SUT. 1. Test setup according to figure Users A, B and C registered. 1 User B calls User A (using the SUT). 2 User A performs an unattended transfer of User B to User C. 3 User B immediately reports to User A that he is trying call User C. 4 User A disconnects with User B. 5 User B establishes the session to User C. V A R I A T I O N S None 1 Speech path established between User B and SUT. 2 SUT sends REFER with header Refer-To to User B. 3 User B sends NOTIFY with header Event: refer to SUT. 4 Speech path between User B and SUT terminates. 5 Speech path established between User B and User C.

64 64 SIP_IOP_SS_CF_001 Call Forwarding Busy User A calls User B (using the SUT), User B is busy. The proxy server places the call to User B. User A terminates the session with User B. To verify behavior of SUT when SUT is in busy state. The proxy server knows the call to User B forwarded to User B if User B is busy 1 User A calls User B (using the SUT). 2 User B is busy 3 The proxy server forwards the call to User B. 4 User A establishes the session to User B. V A R I A T I O N S None 1 Speech path established between User A and SUT. 2 SUT sends 486 Busy Here 3-a User A receives 181 Call is Being forwarded. 3-b User B receives INVITE 3-c User A receives 180 Ringing 4 Speech path established between User A and User B.

65 65 SIP_IOP_SS_CO_001 3-way Conference User A and User B (using the SUT) establish the session. User B adds another User C into the conversation. Proxy Authentication = required; Record Route = enabled To verify that a conference call can be successfully established from SUT. 1. Test setup according to figure Users A,B and C registered. 3. User B is capable of media mixing in a 3-party call. 1 User A establishes a session with User B (using the SUT). 2 User B wishes to add User C into the conversion. 3 Users A, B and C are all in conference call session. V A R I A T I O N S None 1 2-way (User A, B) speech path established. 2 SUT first sends a re-invite to User A, changing the Contact URI to that indicates SUT as a mixer. 2 SUT then INVITEs User C using the same Contact URI. 3 3-way speech path established.

66 66 6 MOS Mean Opinion Score ITU-T P PESQ Perceptual Evaluation of Speech Quality, ITU-T P.862 PESQ MOS -0.5 ~ 4.5 P MOS 1.02 ~ 4.56 Quality Reference MOS Grade Maximum obtainable for G Excellent 5 Good 4 Fair 3 Poor 2 Bad 1 P.862 MOS Handset cable IP-Phone (SUT) Testing Equipment Phone adapter Phone adapter Network Emulator Audio interface

67 67 Real VoIP Deploy Env. VoIP Traffic Packets A/D, Encoding [G.7xx] Phone Adapter Gateway IP-PBX -Packet Lost(%) -Packet Delay(ms) -Packet Out of Sequence(%) -Jitter (ms) Proxy VoIP Traffic Packets D/A, Decoding [G.7xx] -Packet Lost(%) -Packet Delay(ms) -Packet Out of Sequence(%) -Jitter (ms) Phone Adapter ( )

68 68 packet lost 1%, 2%, 3%, 5%, 10%, 15% Real VoIP Deploy Env. VoIP Traffic Packets Gateway IP-PBX Proxy VoIP Traffic Packets Collect Network Parameters -Packet Lost, Packet Delay, Out of Sequence, Jitter Lab. Env. Reference Parameters A/D, Encoding [G.7xx] Phone Adapter Network Emulator D/A, Decoding [G.7xx] Phone Adapter ( )

69 69 VQ-Test IP Phone IP Phone None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S None None

70 70 VQ -Test IP Phone Wi-Fi Phone None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S 1 ( MOS FXS RJ-9, ) None

71 71 VQ -Test Wi-Fi Phone Wi-Fi Phone None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S 1 MOS FXS RJ-9 None

72 72 VQ -Test IP Phone PSTN None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S None None

73 73 VQ -Test Wi-Fi Phone PSTN None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S 1 MOS FXS RJ-9 None

74 74 VQ -Test IP Phone IP Phone None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S None None

75 75 VQ -Test IP Phone Wi-Fi Phone None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S 1 MOS FXS RJ-9 None

76 76 VQ -Test Wi-Fi Phone Wi-Fi Phone None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S 1 MOS FXS RJ-9 None

77 77 VQ -Test IP Phone IP PSTN None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S None None

78 78 VQ -Test Wi-Fi Phone IP PSTN None 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S 1 MOS FXS RJ-9 None

79 79 VQ -Test PSTN IP 2M 512 ADSL 1 Packet Lost (%) 2 Packet Delay (ms) 3 Packet Out of Sequence (%) 4 Jitter (ms) 5 MOS V A R I A T I O N S None None

80 80 7 VoIP A. B.

81 Load-Test Proxy Server Proxy Server CPE A makes calls through Proxy B 1 Call Length Terminates (s) 2 Call Setup Time (ms) 3 Tear Down Time (ms) 4 Busy Hour Call Attempt (BHCA) 5 Concurrent Call Setup (s) 6 RTP Packet Lost (per call) RTP RTP Optional 7 RTP Packet out of Sequence RTP RTP Optional 8 RTP Packet Late Arrival RTP RTP Optional 9 RTP Jitter RTP Jitter RTP Optional V A R I A T I O N S None None

82 82 Load-Test Relay Server Relay Server CPE A makes calls to CPE B thru Relay server 1 Call Length Terminates(s) 2 RTP Packet Lost(per call) RTP RTP Optional 3 RTP Packet out of Sequence RTP RTP Optional 4 RTP Packet Late Arrival RTP RTP Optional 5 RTP Jitter RTP Jitter RTP Optional V A R I A T I O N S None None

83 83 Load-Test IP-PBX Server IP-PBX Server IP-PBX A makes calls to IP-PBX B 1 Call Length Terminates (s) 2 Call Setup Time (ms) 3 Tear Down Time (ms) 4 Busy Hour Call Attempt (BHCA) 5 Concurrent Call Setup (s) V A R I A T I O N S None None

84 Load-Test IP-PBX A makes calls to IP-PBX B ( 2M 512 ADSL ) 1 Call Length Terminates (s): 2 Call Setup Time (ms): 3 Tear Down Time (ms): 4 Packet Lost (%): 5 Packet Delay (ms) : 6 Packet Out of Sequence (%): 7 Jitter (ms): 8 Concurrent calls (call): V A R I A T I O N S None None