Skip to content

/AWS1/IF_CNP=>CREATEPARTICIPANTCONNECTION()

About CreateParticipantConnection

Creates the participant's connection.

For security recommendations, see Amazon Connect Chat security best practices.

For WebRTC security recommendations, see Amazon Connect WebRTC security best practices.

ParticipantToken is used for invoking this API instead of ConnectionToken.

The participant token is valid for the lifetime of the participant – until they are part of a contact. For WebRTC participants, if they leave or are disconnected for 60 seconds, a new participant needs to be created using the CreateParticipant API.

For WEBSOCKET Type:

The response URL for has a connect expiry timeout of 100s. Clients must manually connect to the returned websocket URL and subscribe to the desired topic.

For chat, you need to publish the following on the established websocket connection:

{"topic":"aws/subscribe","content":{"topics":["aws/chat"]}}

Upon websocket URL expiry, as specified in the response ConnectionExpiry parameter, clients need to call this API again to obtain a new websocket URL and perform the same steps as before.

The expiry time for the connection token is different than the ChatDurationInMinutes. Expiry time for the connection token is 1 day.

For WEBRTC_CONNECTION Type:

The response includes connection data required for the client application to join the call using the Amazon Chime SDK client libraries. The WebRTCConnection response contains Meeting and Attendee information needed to establish the media connection.

The attendee join token in WebRTCConnection response is valid for the lifetime of the participant in the call. If a participant leaves or is disconnected for 60 seconds, their participant credentials will no longer be valid, and a new participant will need to be created to rejoin the call.

Message streaming support: This API can also be used together with the StartContactStreaming API to create a participant connection for chat contacts that are not using a websocket. For more information about message streaming, Enable real-time chat message streaming in the Amazon Connect Administrator Guide.

Multi-user web, in-app, video calling support:

For WebRTC calls, this API is used in conjunction with the CreateParticipant API to enable multi-party calling. The StartWebRTCContact API creates the initial contact and routes it to an agent, while CreateParticipant adds additional participants to the ongoing call. For more information about multi-party WebRTC calls, see Enable multi-user web, in-app, and video calling in the Amazon Connect Administrator Guide.

Feature specifications: For information about feature specifications, such as the allowed number of open websocket connections per participant or maximum number of WebRTC participants, see Feature specifications in the Amazon Connect Administrator Guide.

The Amazon Connect Participant Service APIs do not use Signature Version 4 authentication.

Method Signature

IMPORTING

Required arguments:

iv_participanttoken TYPE /AWS1/CNPPARTICIPANTTOKEN /AWS1/CNPPARTICIPANTTOKEN

This is a header parameter.

The ParticipantToken as obtained from StartChatContact API response.

Optional arguments:

it_type TYPE /AWS1/CL_CNPCONNTYPELIST_W=>TT_CONNECTIONTYPELIST TT_CONNECTIONTYPELIST

Type of connection information required. If you need CONNECTION_CREDENTIALS along with marking participant as connected, pass CONNECTION_CREDENTIALS in Type.

iv_connectparticipant TYPE /AWS1/CNPBOOL /AWS1/CNPBOOL

Amazon Connect Participant is used to mark the participant as connected for customer participant in message streaming, as well as for agent or manager participant in non-streaming chats.

RETURNING

oo_output TYPE REF TO /aws1/cl_cnpcreparticipantcx01 /AWS1/CL_CNPCREPARTICIPANTCX01

Domain /AWS1/RT_ACCOUNT_ID
Primitive Type NUMC

Examples

Syntax Example

This is an example of the syntax for calling the method. It includes every possible argument and initializes every possible value. The data provided is not necessarily semantically accurate (for example the value "string" may be provided for something that is intended to be an instance ID, or in some cases two arguments may be mutually exclusive). The syntax shows the ABAP syntax for creating the various data structures.

DATA(lo_result) = lo_client->/aws1/if_cnp~createparticipantconnection(
  it_type = VALUE /aws1/cl_cnpconntypelist_w=>tt_connectiontypelist(
    ( new /aws1/cl_cnpconntypelist_w( |string| ) )
  )
  iv_connectparticipant = ABAP_TRUE
  iv_participanttoken = |string|
).

This is an example of reading all possible response values

lo_result = lo_result.
IF lo_result IS NOT INITIAL.
  lo_websocket = lo_result->get_websocket( ).
  IF lo_websocket IS NOT INITIAL.
    lv_presignedconnectionurl = lo_websocket->get_url( ).
    lv_iso8601datetime = lo_websocket->get_connectionexpiry( ).
  ENDIF.
  lo_connectioncredentials = lo_result->get_connectioncredentials( ).
  IF lo_connectioncredentials IS NOT INITIAL.
    lv_participanttoken = lo_connectioncredentials->get_connectiontoken( ).
    lv_iso8601datetime = lo_connectioncredentials->get_expiry( ).
  ENDIF.
  lo_webrtcconnection = lo_result->get_webrtcconnection( ).
  IF lo_webrtcconnection IS NOT INITIAL.
    lo_attendee = lo_webrtcconnection->get_attendee( ).
    IF lo_attendee IS NOT INITIAL.
      lv_attendeeid = lo_attendee->get_attendeeid( ).
      lv_jointoken = lo_attendee->get_jointoken( ).
    ENDIF.
    lo_webrtcmeeting = lo_webrtcconnection->get_meeting( ).
    IF lo_webrtcmeeting IS NOT INITIAL.
      lo_webrtcmediaplacement = lo_webrtcmeeting->get_mediaplacement( ).
      IF lo_webrtcmediaplacement IS NOT INITIAL.
        lv_uri = lo_webrtcmediaplacement->get_audiohosturl( ).
        lv_uri = lo_webrtcmediaplacement->get_audiofallbackurl( ).
        lv_uri = lo_webrtcmediaplacement->get_signalingurl( ).
        lv_uri = lo_webrtcmediaplacement->get_eventingestionurl( ).
      ENDIF.
      lo_meetingfeaturesconfigur = lo_webrtcmeeting->get_meetingfeatures( ).
      IF lo_meetingfeaturesconfigur IS NOT INITIAL.
        lo_audiofeatures = lo_meetingfeaturesconfigur->get_audio( ).
        IF lo_audiofeatures IS NOT INITIAL.
          lv_meetingfeaturestatus = lo_audiofeatures->get_echoreduction( ).
        ENDIF.
      ENDIF.
      lv_guidstring = lo_webrtcmeeting->get_meetingid( ).
    ENDIF.
  ENDIF.
ENDIF.