Dialogic Global Call IP - Manual

Dialogic Global Call IP

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Table of Contents:

  • Page 3 – Contents; Dialogic
  • Page 4 – IP-Specific Operations
  • Page 7 – Third Party Call Control (3PCC) Operations and Multimedia Support
  • Page 8 – IP-Specific Function Information
  • Page 9 – IP-Specific Parameters
  • Page 10 – IP-Specific Event Cause Codes
  • Page 11 – Figures
  • Page 13 – Tables
  • Page 15 – Revision History
  • Page 25 – About This Publication; Purpose; Applicability
  • Page 26 – How to Use This Publication
  • Page 29 – IP Overview; Introduction to VoIP; Codecs
  • Page 32 – Provides a bit rate of 64 kbps.; Basic H.323 Call Scenario; A simple H.323 call scenario can be described in five phases:
  • Page 33 – Call Setup
  • Page 34 – Call Initiation; Section 4.2, “Fast and Slow Call Setup Modes”; Data Exchange
  • Page 35 – Registration with a Gatekeeper; admission control for authorizing network access; Gatekeeper Discovery; An endpoint uses a process called
  • Page 36 – Endpoint Registration; Endpoint Deregistration; H.323 Call Scenario via a Gateway; While the call scenario described in
  • Page 37 – Figure 3. Basic H.323 Network with a Gateway; The call scenario in this example involves the following phases:; Establishing Contact with the Gatekeeper; a sequence number
  • Page 38 – Call Signaling and Data Exchange; Capability Exchange; Call Termination
  • Page 39 – SIP Overview; Advantages of Using SIP; Some of the advantages of using SIP include:; SIP User Agents and Servers; Location Server
  • Page 40 – Basic SIP Operation; Figure 4. Basic SIP Call Scenario; SIP Messages; SIP Request Messages
  • Page 44 – Global Call API Over IP Architecture; Architecture Components; Host
  • Page 45 – Host Application
  • Page 46 – IP Media Call Control Library (IPM CCLib); Device Types and Usage; Device Types Used with IP; iptBx
  • Page 47 – Global Call API Devices; IPT Board Devices
  • Page 48 – IPT Network Devices; devicename
  • Page 49 – IPT Start Parameters
  • Page 51 – This chapter describes how to use the Dialogic
  • Page 52 – Call Control Library Initialization
  • Page 53 – Setting a SIP Outbound Proxy; function. If both an IP address and a host
  • Page 54 – Configuring SIP Transport Protocol; When TCP is enabled, the Dialogic
  • Page 55 – Configuring TCP Transport
  • Page 56 – Transport Parameter Combinations without Proxy; All Requests UDP
  • Page 57 – Selected Requests UDP; Transport Parameter Combinations with Proxy; All Requests UDP via Proxy
  • Page 58 – Selected Requests TCP via Proxy; Invalid Transport Parameter Combinations; TCP is not enabled
  • Page 59 – cannot; Fast and Slow Call Setup Modes; Setting the Call Setup Mode
  • Page 60 – duration; H.323 Fast Start and Slow Start; fastStart
  • Page 61 – H.323 Fast Start with Optional H.245 Channel
  • Page 62 – SIP Call Setup Modes
  • Page 63 – Retrieving Coder Information from Call Offers; Enabling Access to “Fast Start” Coder Information
  • Page 64 – Accessing “Fast Start” Coder Information; Information”
  • Page 65 – Setting Call-Related Information; The Dialogic
  • Page 66 – Overview of Setting Call-Related Information; Table 1. Summary of Call-Related Information that can be Set
  • Page 68 – Setting Call Parameters on a System-Wide Basis; Setting Call Parameters on a Per Call Basis
  • Page 69 – Setting Coder Information
  • Page 71 – Specifying Media Capabilities Before Connection; Table 2. Coders Supported for Dialogic
  • Page 72 – Resource Allocation When Using Low-Bit Rate Coders
  • Page 75 – Connection Phase Messages; Setting and Retrieving Disconnect Cause or Reason Values
  • Page 76 – cause; Setting Busy Reason Codes; Setting SIP Busy Code
  • Page 77 – Setting H.323 Busy Code
  • Page 78 – first; 82 Queued response for a given call.
  • Page 79 – Retrieving Status-Code for 18x Provisional Responses; Section 4.9, “Setting and Retrieving SIP Message Header Fields”
  • Page 80 – Redirecting an Incoming Call
  • Page 82 – Receiving and Handling a Redirect Response
  • Page 84 – SIP Rejection Responses; The information in this section only applies when the Dialogic
  • Page 85 – Retrieving Current Call-Related Information
  • Page 86 – parmblkp
  • Page 87 – Table 4. Retrievable Call Information
  • Page 89 – Examples of Retrieving Call-Related Information; extID; Table 4, “Retrievable Call Information”; Specifying Call-Related Information to Retrieve
  • Page 92 – Event
  • Page 95 – Retrieving Call ID
  • Page 96 – Parsing Call ID Information (SIP Protocol); call when the SIP protocol is being used.
  • Page 98 – Receiving Notification Events; Enabling and Disabling Unsolicited Notification Events; Status and Connection Information”; size
  • Page 100 – RTP Address and Coder Information Retrieval Example
  • Page 101 – Getting Notification of Underlying Protocol State Changes; Disabling Unsolicited Notification Events”; Modifying an Existing SIP Call via re-INVITE; This section discusses the Dialogic
  • Page 102 – Overview of the SIP re-INVITE Method; Global Call API library for Dialogic
  • Page 103 – Enabling Application Access to re-INVITE Requests
  • Page 106 – Responding to SIP re-INVITE Requests; Rejecting a SIP re-INVITE Request; function to; Accepting a SIP re-INVITE Request; If the application calls
  • Page 107 – Sending a SIP re-INVITE Request; produces
  • Page 108 – Specifying DTMF Mode in a re-INVITE Request; value = complete header string, including header field name
  • Page 109 – Specifying Media Session Properties in a SIP re-INVITE; The parameter elements for media capabilities are of the form:; Canceling a Pending re-INVITE Request; The parameter element used to cancel a pending re-INVITE is:
  • Page 110 – Updating Dialog Properties via re-INVITE
  • Page 111 – Implementing Hold and Retrieve via SIP re-INVITE; Suspending RTP Streaming Only; . The parameter element conforms to the; Suspending RTP and RTCP Streaming; parameter element conforms to the following:
  • Page 112 – Retrieving a Held Call
  • Page 113 – Setting and Retrieving Q.931 Message IEs; Enabling Access to Q.931 Message IEs; not
  • Page 114 – Supported Q.931 Message IEs; outgoing; infoparmblkp; Retrieving Q.931 Message IEs; incoming; Table 5. Supported Q.931 Message Information Elements
  • Page 115 – Common Usage Scenarios Involving Q.931 Message IEs; Table 7 shows how the Dialogic; Table 6. Supported IEs in Incoming Q.931 Messages
  • Page 116 – Setting and Retrieving SIP Message Header Fields; SIP Header Access Overview; Header Fields in Outgoing SIP Messages
  • Page 118 – Table 8. Common Header Fields in Outbound SIP Messages (Continued)
  • Page 119 – Header Fields in Incoming SIP Messages; For incoming SIP messages, the Dialogic
  • Page 120 – Table 9. Common Header Fields in Inbound SIP Messages (Continued)
  • Page 121 – API Functions for Long Header Values; Global Call API; Field-Specific Parameters for SIP Header Access; must
  • Page 123 – Enabling Access to SIP Header Information; Enabling Long Header Values
  • Page 124 – Registering SIP Header Fields to be Retrieved
  • Page 127 – Setting SIP Header Fields for Outbound Messages; Section 4.9.4, “Registering SIP Header Fields to be Retrieved”; Table 8, “Common Header Fields in Outbound SIP Messages”
  • Page 129 – Retrieving SIP Message Header Fields; “Enabling Access to SIP Header Information”
  • Page 132 – SIP MIME Overview
  • Page 133 – use the extended; Figure 47. SIP MIME Scenario for Normal Call Setup and Teardown
  • Page 134 – Figure 48. SIP MIME Scenario for Rejected Call
  • Page 139 – Code Example
  • Page 141 – Sending MIME Information; and
  • Page 142 – Create MIME part structures
  • Page 143 – Set/send message data and clean up; or; because the Content-Type and Content-Disposition header strings
  • Page 144 – MIME Error Conditions
  • Page 145 – Specifying Transport for SIP Messages; Protocol”
  • Page 146 – Handling SIP Transport Failures; configuration enables all allowable retries.
  • Page 149 – Sending and Receiving SIP INFO Messages; Global Call API library provides facilities; Sending an INFO Message; Section 4.9.5, “Setting SIP Header Fields for Outbound Messages”
  • Page 150 – Receiving a Response to an INFO Message
  • Page 151 – call is
  • Page 152 – Receiving an INFO Message; Message Header Fields”
  • Page 153 – Responding to an INFO Message; Global Call API library fills in the default value 200.; “OK” Response to INFO Message
  • Page 154 – Sending and Receiving SIP OPTIONS Messages; The following topics describe the Dialogic; Default OPTIONS Behavior; Global Call API library responds to incoming OPTIONS requests
  • Page 155 – Enabling Application Access to OPTIONS Messages
  • Page 156 – SIP Header Fields to be Retrieved”; Sending OPTIONS Requests
  • Page 157 – “Setting SIP Header Fields for Outbound Messages”
  • Page 159 – Receiving Responses to OPTIONS Requests
  • Page 160 – Contact URI
  • Page 161 – Receiving OPTIONS Requests; Fields”
  • Page 162 – Responding to OPTIONS Requests; use the; “Success” Response Message
  • Page 164 – “Failed” Response Message; When sending the response message, the application; include the Call-ID header field value
  • Page 166 – Using SIP SUBSCRIBE and NOTIFY Messages; with Ringing”
  • Page 167 – Sending SUBSCRIBE Requests; function to send the message. The header may include any
  • Page 169 – Receiving Responses to SUBSCRIBE Requests
  • Page 172 – Receiving SUBSCRIBE Requests
  • Page 173 – “Receiving Responses to SUBSCRIBE Requests”; Responding to SUBSCRIBE Requests; include the Call-ID header field value that was received in the; “Accept” response to SUBSCRIBE request
  • Page 174 – “Reject” response to SUBSCRIBE request
  • Page 175 – Sending NOTIFY Requests
  • Page 178 – Receiving Responses to NOTIFY Requests
  • Page 179 – Receiving NOTIFY Requests; Responding to NOTIFY Requests
  • Page 180 – “Accept” Response to NOTIFY Request
  • Page 181 – “Reject” Response to NOTIFY Request; Handling DTMF; DTMF handling is described under the following topics:
  • Page 182 – Generating or Detecting DTMF Tones Using a Voice Resource; Specifying DTMF Support; Bitmask values for SIP; Bitmask values for H.323
  • Page 184 – any value in the range 96 to 127 – (dynamic payload type; Getting Notification of DTMF Detection; Once DTMF support has been configured (see; Table 14. Summary of DTMF Mode Settings and Behavior
  • Page 185 – and Disabling Unsolicited Notification Events”; Generating DTMF
  • Page 187 – “IPSET_NONSTANDARDDATA”
  • Page 189 – Nonstandard Registration Message
  • Page 190 – Sending Facility, UII, or Registration Message Scenario
  • Page 191 – Figure 50. Sending Protocol Messages; Using H.323 Annex M Tunneled Signaling Messages; Tunneled Signaling Message Overview; Global Call API library does not support
  • Page 192 – Tunneled Signaling Messages”
  • Page 193 – Enabling Tunneled; Enabling Tunneled Signaling Messages
  • Page 194 – Composing Tunneled Signaling Messages; one
  • Page 196 – Sending Tunneled Signaling Messages
  • Page 197 – Sending a TSM in a Setup Message; ReleaseComplete Message
  • Page 198 – Receiving Tunneled Signaling Messages
  • Page 199 – ReleaseComplete, and Setup Messages
  • Page 201 – TSM Retrieval Code Example
  • Page 203 – Enabling Reception of User-to-User Information; Table 18. H.225 Messages and Global Call Events for Receiving UU-IE
  • Page 204 – method and the
  • Page 205 – Retrieving UU-IE from Facility Messages
  • Page 206 – Specifying RTP Stream Establishment
  • Page 207 – Managing Quality of Service Alarms; Global Call API Programming Guide
  • Page 208 – Alarm Source Object Name; IPM QoS ASO; Retrieving the Media Device Handle
  • Page 209 – Retrieving QoS Threshold Values; To retrieve QoS threshold values, use the
  • Page 210 – Handling QoS Alarms
  • Page 211 – When a GCEV_ALARM event occurs, use the Dialogic
  • Page 212 – Registration; Registration Overview; H.323 Gatekeeper Registration
  • Page 213 – SIP Registration; bindings
  • Page 214 – binding exists between the AOR; in the; Table 19. SIP REGISTER Method
  • Page 215 – Registration Operations; Configuring the Maximum Number of Registrations (SIP)
  • Page 216 – function is called and remain in effect until; Locating a Registration Server; A Dialogic; Registration Requests; An application uses the
  • Page 217 – SIP
  • Page 218 – Global Call API library only if the application
  • Page 219 – Receiving Notification of Registration; Global Call API, the; was made
  • Page 220 – all; Changing Registration Information
  • Page 221 – Table 38, “Registration Information When Using; Removing Registered Information by Value
  • Page 222 – Deregistering
  • Page 223 – Registration Code Examples
  • Page 224 – Registration Example
  • Page 227 – Deregistration Example; two mandatory parameters required by the generic
  • Page 229 – SIP Digest Authentication
  • Page 232 – Overview of TLS
  • Page 234 – digital certificate
  • Page 235 – An example of an RSA certificate in PEM format is shown below.
  • Page 236 – Certificate Authority
  • Page 239 – Certificate Revocation List
  • Page 240 – Configuring and Enabling TLS; is called
  • Page 241 – , the library start operation will fail.; Changing the Default TLS Port Number
  • Page 242 – Configuring CA Certificates for TLS Client Operation
  • Page 243 – Configuring Server Session Caching; Note that; Setting the Mutual Authentication Option; Global Call API library; Setting the Insecure Port Blocking Options
  • Page 244 – Simple SIP_TLS_ENGINE Configuration Example
  • Page 245 – Enabling TCP in IP_VIRTBOARD; initialization will fail.
  • Page 246 – Configuring TCP/TLS Persistence in IP_VIRTBOARD; Transport Protocol”; Enabling TLS in IP_VIRTBOARD; The library will also fail to load when
  • Page 247 – Making Calls Using TLS; Outbound proxy transport configured to be TLS
  • Page 249 – Source address is “sip:” URI, destination address is “sip:” URI
  • Page 250 – Figure 53. TLS with “sip:” Source Address and “sip:” Destination Address
  • Page 251 – Source address is “sip:” URI, destination address is “sips:” URI
  • Page 252 – Source address is “sips:” URI, destination address is “sip:” URI
  • Page 253 – Source address is “sips:” URI, destination address is “sips:” URI
  • Page 254 – TLS Transport Failures; Call Transfer
  • Page 255 – Enabling Call Transfer
  • Page 256 – Figure 58. Global Call Devices for Supervised Call Transfer; Incoming Transferred Call; Global Call API library (GC LIB) cause value of
  • Page 257 – Call Transfer Glare Condition; to
  • Page 258 – Figure 59. Call Transfer Glare Condition
  • Page 259 – Call Transfer When Using SIP; page 74. The topics covered here include:; Invoking an Unattended Call Transfer
  • Page 260 – Invoking an Attended Call Transfer; both; parties before; Processing Asynchronous Call Transfer Events
  • Page 262 – Handling a Transfer Request
  • Page 263 – Making a Transferred Call
  • Page 264 – T.38 Fax Server Support Overview
  • Page 265 – For SIP, the association must be made before; optionally; reestablishing the audio connection,; before; the; Figure 60. T.38 Fax Server Support in Manual Mode
  • Page 266 – Specifying Manual Operating Mode; function
  • Page 267 – function. To continue to accept the request, the
  • Page 269 – When a Fax Request is Received”; Sending a T.38 Fax in a Session Without Audio Established
  • Page 270 – only
  • Page 271 – function. The following code
  • Page 272 – Sending a Request to Switch from T.38 Fax to Audio; function. The application
  • Page 273 – Receiving a Request to Switch from T.38 Fax to Audio; continue accepting the request as described in
  • Page 274 – Terminating a Call After a T.38 Fax Session; Sending and Receiving V.17 Faxes; Sending G.711 Fax in an Established Audio Session
  • Page 275 – Figure 61. Sending G.711 Fax in an Established Audio Session; Receiving G.711 Fax in an Established Audio Session; Global Call API to open
  • Page 276 – Figure 62. Receiving G.711 Fax in an Established Audio Session; Using Object Identifiers
  • Page 277 – LAN Disconnection Alarms
  • Page 278 – Explicitly open the board device.
  • Page 279 – Setting Dialogic
  • Page 281 – IP Call Scenarios; Global Call API with; Basic Call Control Scenarios When Using; IP Technology; Basic Call Setup When Using H.323 or SIP
  • Page 282 – Figure 8 shows the basic call setup sequence when using H.323 or SIP.; Figure 8. Basic Call Setup When Using H.323 or SIP
  • Page 283 – Basic Call Teardown When Using H.323 or SIP; Global Call API with H.323; Figure 9. Basic Call Teardown When Using H.323 or SIP; Call Setup Scenarios for Early Media; early media
  • Page 285 – SIP FastStart Mode (Calling UA Offers SDP)
  • Page 286 – Figure 12. SIP Early Media, Calling UA Offers SDP; SIP SlowStart Mode (Calling UA Answers SDP); delayed offer; Figure 13. SIP Early Media, Calling UA Answers SDP
  • Page 287 – Call Transfer Scenarios When Using H.323; General Conditions for H.450.2 Call Transfers; Global Call API, party A and party B are
  • Page 289 – Successful H.450.2 Blind Call Transfer Scenario; Global Call API. Completion of a
  • Page 291 – Unsuccessful H.450.2 Blind Call Transfer Scenarios; Party B Rejects Transfer
  • Page 292 – No Response From Party B
  • Page 293 – No Response From Party C
  • Page 294 – Party B Clears Primary Call Before Transfer is Completed; Before Transfer is Completed
  • Page 295 – Party C is Busy When Transfer Attempted
  • Page 296 – Endpoint Behavior in H.450.2 Supervised Call Transfer; function at; reroutinginfop
  • Page 297 – Successful H.450.2 Supervised Call Transfer Scenario; successful
  • Page 299 – Unsuccessful H.450.2 Supervised Transfer Scenarios; Party C Timeout
  • Page 301 – Party C Rejects the Transfer Request
  • Page 302 – Party B Rejects the Transfer Request
  • Page 303 – Party B Timeout
  • Page 304 – Call Transfer Scenarios When Using SIP; General Conditions for SIP Call Transfers; There are two fundamental types of call transfer:
  • Page 305 – Interoperability Issues; no initial NOTIFY after sending out 202 accept to REFER request; Endpoint Behavior in Unattended SIP Call Transfers; Transferor or Transferring Endpoint (party A)
  • Page 306 – Transferee or Transferred Endpoint (Party B)
  • Page 308 – Successful Unattended SIP Call Transfer Scenarios; Successful Transfer with Notification of Connection
  • Page 310 – Successful Transfer with Notification of Ringing
  • Page 311 – prior to Notification of Transferred Call Status
  • Page 312 – Transfer Completion
  • Page 314 – Endpoint Behavior in Attended SIP Transfers; Transferor or Transferring Endpoint (Party A); function. The difference between unattended and attended; Protecting and Exposing the Transfer Target
  • Page 316 – Successful SIP Attended Call Transfer Scenarios; Successful SIP Attended Call Transfer
  • Page 317 – function call, which is not required in SIP. The; Figure 30. Successful SIP Attended Call Transfer
  • Page 318 – Attended Transfer when REFER is Not Globally Supported
  • Page 320 – Unsuccessful Call Transfer Scenarios; function call and the corresponding; Party B Rejects Call Transfer; to signal the Transferor (party A) that it cannot participate in a
  • Page 321 – Figure 33. SIP Call Transfer Failure - Party B Rejects Call Transfer; Figure 34. SIP Call Transfer Failure - No Response from Party B
  • Page 322 – No Initial NOTIFY after REFER Accepted
  • Page 323 – REFER Subscription Expires; Figure 36. SIP Call Transfer Failure - REFER Subscription Expires
  • Page 324 – Figure 37. SIP Call Transfer Failure - No Response from Party C
  • Page 325 – Party B Drops Transferred Call Early
  • Page 327 – T.38 Fax Server Call Scenarios
  • Page 328 – Sending T.38 Fax in an Established Audio Session; Figure 40. Sending T.38 Fax in an Established Audio Session
  • Page 329 – Receiving T.38 Fax in an Established Audio Session
  • Page 330 – Figure 41. Receiving T.38 Fax in an Established Audio Session
  • Page 331 – Sending T.38 Fax Without an Established Audio Session; Figure 42. Sending T.38 Fax Without an Established Audio Session
  • Page 332 – Receiving T.38 Fax Without an Established Audio Session; The application must not use both Dialogic; Figure 43. Receiving T.38 Fax Without an Established Audio Session
  • Page 333 – Sending a Request to Switch From T.38 Fax to Audio; Figure 44. Sending a Request to Switch From T.38 Fax to Audio
  • Page 334 – Receiving a Request to Switch From T.38 Fax to Audio; Figure 45. Receiving a Request to Switch From T.38 Fax to Audio
  • Page 335 – Recovering from a Session Switching Failure
  • Page 337 – Global Call API to perform certain third party call; Overview; Third Party Call Control; The key attributes of third party call control are:
  • Page 338 – Figure 63. Third Party Call Controller
  • Page 339 – Figure 64. Basic Call Setup When Using Third Party Call Control; The call sequence description for Figure 64 is as follows:
  • Page 340 – Figure 65. Third Party Call Control Setup using re-INVITE; The call sequence description for Figure 65 is as follows:
  • Page 342 – Session Description Protocol
  • Page 343 – Global Call in Third Party Call Control Mode; Initializing the Library in Third Party Call Control Mode
  • Page 344 – Interface Changes; IPSET_SDP Parameter Set Identifier
  • Page 345 – Model with Session Description Protocol (SDP)
  • Page 348 – IPSET_SDP Code Example
  • Page 350 – Events; Table 23. Global Call Third Party Call Control Mode Events
  • Page 351 – Global Call Functions Invalid in Third Party Call Control Mode; When the Dialogic; Table 24. Global Call Third Party Call Control Mode Error Codes
  • Page 352 – Call Setup Modes”; Call Transfer Scenarios; Section 3.3, “Call Transfer Scenarios When Using SIP”
  • Page 353 – DTMF Transport; Message Sequence Diagrams; First Party Call Establishment in Third Party Call Control Mode
  • Page 355 – Basic Third Party Call Control Establishment
  • Page 356 – Figure 68. Basic Third Party Call Control Establishment (part one)
  • Page 357 – Figure 69. Basic Third Party Call Control Establishment (part two)
  • Page 358 – Alternate Third Party Call Control Establishment
  • Page 362 – Modifying the Coder; Successfully Modifying the Coder
  • Page 365 – Unsuccessfully Modifying the Coder; Figure 75. Unsuccessfully Modifying the Coder
  • Page 366 – Cancelling a re-INVITE Request; Figure 76. Cancelling a Coder Switch using re-INVITE
  • Page 367 – Receiving an Invalid Answer SDP; determining that the received answer SDP is invalid
  • Page 368 – Figure 77. Receiving an Invalid Answer SDP; OPTIONS Request on an Active Dialog; OPTIONS Request Without a MIME Body
  • Page 369 – Refer to
  • Page 370 – Figure 78. OPTIONS Request without a MIME Body; With a MIME Body
  • Page 371 – Figure 79. OPTIONS Request with a MIME Body; Processing Intraframe Requests for Video Streams
  • Page 372 – Requesting an I-Frame in SIP; . The draft specification defines an XML; Global Call Example Code
  • Page 375 – Building Dialogic; Header Files
  • Page 376 – Required System Software
  • Page 377 – Debugging Dialogic; Debugging Overview
  • Page 378 – Configuring the Logging Facility; Configuration File Overview; Global Section
  • Page 379 – Module Sections; The; Client Entries; Configuring the gc_h3r Logging Module; file in previous implementations.
  • Page 382 – Configuring SIP Stack Logging
  • Page 383 – Configuring H.323 Stack Logging
  • Page 394 – Global Call API Functions
  • Page 396 – accept proposed modification of call characteristics; Description
  • Page 397 – with a NULL pointer as
  • Page 398 – as an; Termination Events; Cautions
  • Page 399 – Errors; Example
  • Page 401 – The following code example illustrates how the
  • Page 405 – See Also
  • Page 406 – reject proposed modification of call attributes; Section 11.5, “Failure Response Codes When Using SIP”
  • Page 414 – gc_ReqModifyCall( ) — request modification of call attributes
  • Page 415 – request modification of call attributes — gc_ReqModifyCall( )
  • Page 416 – in 3PCC mode; , a call control application; . If the response code from the remote party was a 408 Request
  • Page 421 – set IP authentication information — gc_SetAuthenticationInfo( )
  • Page 422 – gc_SetAuthenticationInfo( ) — set IP authentication information; Examples
  • Page 424 – acknowledge a SIP 200OK message in 3PCC mode
  • Page 427 – None
  • Page 433 – insert a GC_PARM_BLK parameter by reference
  • Page 436 – retrieve the next parameter in a GC_PARM_BLK
  • Page 440 – reference page
  • Page 441 – initialize IP_VIRTBOARD data structure
  • Page 444 – only in asynchronous mode; rings; Variance for SIP
  • Page 445 – Global Call API cause value of
  • Page 449 – “IP-Specific Event Cause Codes”
  • Page 453 – valuep
  • Page 454 – Returns the simple destination address in the form; argument contains a
  • Page 457 – Table 28. H.450.2 ctInitiate Errors Received from the Network
  • Page 458 – function as
  • Page 459 – Section 4.9, “Setting and Retrieving SIP Message Header
  • Page 460 – Table 33. SIP Header Fields Settable in REFER Messages
  • Page 461 – Configurable Call Parameters
  • Page 464 – Origination Address Information; The origination address can be set using the; Table 35. Configurable Call Parameters When Using SIP
  • Page 465 – Forming a Destination Address String; numberstr; Table 34, “Configurable Call Parameters When Using H.323”
  • Page 466 – Table 35, “Configurable Call Parameters When Using SIP”
  • Page 467 – Scenario 2
  • Page 468 – Destination Address Interpretation
  • Page 469 – Scenario 3
  • Page 470 – Scenario 6; Specifying a Timeout; timeout; Variances for IP”
  • Page 471 – Code Examples
  • Page 474 – SIP-Specific Code Example
  • Page 477 – dev
  • Page 478 – reason
  • Page 480 – reqdatap; See; Table 38. Registration Information When Using H.323
  • Page 481 – function is shown in Table 39.
  • Page 482 – ignored; This function is only supported in asynchronous mode.; Table 39. Registration Information When Using SIP
  • Page 483 – datap
  • Page 485 – for IP”
  • Page 488 – Table 34, “Configurable; Setting Call Parameters for the Next Call
  • Page 489 – Setting Call Parameters for the Next and Subsequent Calls; Setting SIP Message Information Fields
  • Page 491 – “IPT Board Devices”
  • Page 492 – any
  • Page 493 – on page 553 for more; Default configuration parameter values; . If this configuration is not appropriate, or if the application
  • Page 496 – GCEV_EXTENSIONCMPLT [termination event for
  • Page 497 – GCEV_SERVICERESPCMPLT
  • Page 499 – Overview of Parameter Usage
  • Page 500 – Table 42. Summary of Parameter Sets and Parameter Usage (Continued)
  • Page 510 – Parameter Set Reference; The parameter sets documented in this section include:
  • Page 525 – Table 59. IPSET_NONSTANDARDCONTROL Parameter Set
  • Page 526 – IPSET_NONSTANDARDDATA; Table 60. IPSET_NONSTANDARDDATA Parameter Set
  • Page 529 – use the “extended”
  • Page 534 – Table 70. IPSET_TRANSACTION Parameter Set
  • Page 535 – IPSET_TUNNELEDSIGNALMSG; Table 71. IPSET_TUNNELEDSIGNALMSG Parameter Set
  • Page 537 – IP-Specific Data Structures; These data structures are defined in the
  • Page 538 – retrieved parameter data; Dialogic® Global Call API Library Reference
  • Page 540 – IP_ADDR — local IP address; The IP_ADDR structure is used to specify a local IP address.; Field Descriptions; The fields of the IP_ADDR data structure are described as follows:
  • Page 541 – basic audio capability information — IP_AUDIO_CAPABILITY
  • Page 542 – IP_AUTHENTICATION — SIP digest authentication data
  • Page 543 – basic capability information — IP_CAPABILITY; capability
  • Page 544 – IP_CAPABILITY — basic capability information
  • Page 545 – parameters for different capability categories; audio
  • Page 546 – IP_CONNECT — associate a Media device with a T.38 Fax device
  • Page 547 – DTMF information; The DTMF digit string buffer 32 characters in size
  • Page 548 – IP_DATA_CAPABILITY — basic data capability information; The recommended value for T.38 coders is 14400.
  • Page 549 – H.221 nonstandard data — IP_H221NONSTANDARD; extension
  • Page 550 – IP_REGISTER_ADDRESS — gatekeeper registration information; The meaning is protocol dependent:
  • Page 551 – TSM protocol alternate ID — IP_TUNNELPROTOCOL_ALTID
  • Page 552 – tunneled signaling protocol object ID
  • Page 553 – information about an IPT board device — IP_VIRTBOARD
  • Page 554 – IP_VIRTBOARD — information about an IPT board device
  • Page 558 – IP call control library configuration information
  • Page 560 – REQUEST_ERROR — SIP request retry info; version; be modified by applications.
  • Page 561 – RTP address — RTP_ADDR
  • Page 562 – SIP_TLS_ENGINE — TLS engine configuration information
  • Page 563 – TLS engine configuration information — SIP_TLS_ENGINE
  • Page 565 – IP-Specific Error Codes
  • Page 569 – Error Codes When Using H.323
  • Page 570 – For
  • Page 574 – Internal Disconnect Reasons
  • Page 582 – RAS Failure Reasons
  • Page 584 – Failure Response Codes When Using SIP
  • Page 589 – SIP Message Error Codes; MIME memory pool is exhausted.; SIP Registration Error Codes; The SIP stack ran out of resources to process request.
  • Page 591 – Supplementary Reference; References to More Information; The following publications provide related information:
  • Page 592 – Supplementary Reference Information; SIP Transaction Timer Values
  • Page 593 – DNS Configuration for SIP
  • Page 594 – Called Party Address List; Called party address lists are formatted as follows:
  • Page 595 – Calling Party Address List; Called Party Address
  • Page 596 – Examples of Called and Calling Party Addresses; Some examples of called party and calling party addresses are:
  • Page 597 – Glossary; unsupervised transfer
  • Page 598 – and software that initiate and receive calls over a SIP network.; unsupervised transfer:; the person transferring the call.
  • Page 599 – Index; Numerics
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Dialogic

®

Global Call IP

Technology Guide

November 2007

05-2239-009

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Summary

Page 3 - Contents; Dialogic

Dialogic ® Global Call IP Technology Guide – November 2007 3 Dialogic Corporation Contents Revision History . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 About This Publication . . . . . . . . . . . . . . . . . . . . . . ...

Page 4 - IP-Specific Operations

4 Dialogic ® Global Call IP Technology Guide – November 2007 Dialogic Corporation Contents 3.2.2 Endpoint Behavior in H.450.2 Blind Call Transfers . . . . . . . . . . . . . . . . . . . . . . . . 57 3.2.3 Successful H.450.2 Blind Call Transfer Scenario . . . . . . . . . . . . . . . . . . . . . . . . ...

Page 7 - Third Party Call Control (3PCC) Operations and Multimedia Support

Dialogic ® Global Call IP Technology Guide – November 2007 7 Dialogic Corporation Contents 4.26.2 Specifying Manual Operating Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3224.26.3 Initiating a Switch from Audio to T.38 Fax . . . . . . . . . . . . . . . . . . . . . . . . ...

Other Dialogic Models

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