Release Notes

Version 8.0.4 - June 16, 2021

Fixes

• Fix bug on DTMF dispatch of callback to event_url

Version 8.0.3 - May 18, 2021

Fixes

• Fix out of order internal events processing in the events queue
• Fix deprecation warning message for rtcstats:report

Version 8.0.1 - April 29, 2021

Fixes

• Update Typescript definitions
• Fix bug on IP-IP call scenario involving Native SDKs (not populating properly the Conversation.members Map)

Version 8.0.0 - April 27, 2021

Breaking changes

• Deprecate Conversation.members Map (it will be populated only on a call scenario)
• Conversation events will be emitted with a subset information of the Member

conversation.on("any:event", ({memberId, userId, userName, displayName, imageUrl, customData}, event) => {});

Changes

• Add paginated getMembers() function to retrieve the members of a conversation

conversation.getMembers().then((members_page) => {
  members_page.items.forEach(member => {
    render(member);
  })
}).catch((error) => {
  console.error("error getting the members ", error);
});

• Add getMyMember() function to retrieve our own member in a conversation

conversation.getMyMember().then((member) => {
  render(member);
}).catch((error) => {
  console.error("error getting my member", error);
});

• Add getMember() function to fetch a conversation member by member_id

conversation.getMember("MEM-id").then((member) => {
  render(member);
}).catch((error) => {
  console.error("error getting member", error);
});

Version 7.1.0 - April 07, 2021

Changes

• rtcstats:report is deprecated. Please use rtcstats:analyticsinstead

application.on('rtcstats:analytics', ({
  type: 'mos' || 'mos_report',
  mos: string,
  rtc_id: string,
  mos_report?: {
    average: string,
    last: string,
    max: string,
    min: string
  },
  report?: RTCStatsReport
}) => {}

• Add logs reporter for remote logging
• Add custom getStats parser
• Remove callstats library and implementation

Version 7.0.2 - March 30, 2021

Fixes

• Emit 'call:status:update' event when application offline and call disconnected

Version 7.0.1 - February 04, 2021

Fixes

• Remove unnecessary warning message for call status transitions

Version 7.0.0 - February 02, 2021

Breaking changes

• legs endpoint should be included in acl paths on JWT token creation

"acl": {
  "paths": {
    ...
    "/*/legs/**": {}
  }
}

Changes

• Improve callServer setup time by pre-warming leg
• Add the rtcObject and remote stream to the NxmCall object

Fixes

• Return ClientDisconnected reason when client logouts from SDK

Version 6.2.1 - December 24, 2020

Fixes

• Revert back to 6.1.1 from 6.2.0-alpha

Version 6.1.2 - December 24, 2020

Fixes

• Update Typescript definitions

Version 6.1.1 - December 09, 2020

Fixes

• Update Typescript definitions

Version 6.1.0 - December 01, 2020

Changes

• Add enableEventsQueue flag to client configuration (default to true)
• Add internal eventsQueue mechanism to guarantee order of events received during a session

Version 6.0.19 - November 26, 2020

Changes

• Update reconnectionDelay to 2000 in socket.io
• Add randomizationFactor to 0.55 in socket.io

Version 6.0.18 - November 19, 2020

Changes

• Update documentation

Fixes

• Add reconnectionDelay to 3000 and reconnectionDelayMax to 15000 in socket.io

Version 6.0.17 - November 12, 2020

Fixes

• Update Session id after reconnection

Version 6.0.16 - October 12, 2020

Fixes

• Dispatch system:error:expired-token event in application level

  application.on('system:error:expired-token', 'NXM-errors', () => {
    console.log('Token Expired');
  });

• Handle uncaught exception of conversation:error:invalid-member-state event

Version 6.0.15 - September 21, 2020

Fixes

• Set SDK default to single preset ICE candidate
• Remove iceGatherOnlyOneCandidate flag from client configuration

Version 6.0.13 - September 14, 2020

Fixes

• Update websocket reconnection logic for token expiry

Version 6.0.12 - September 01, 2020

Fixes

• Update resolved stream on media enabling

Version 6.0.11 - September 01, 2020

Changes

• Update websocket reconnection logic
• Fix delay in establishing media connection with one ice candidate sent

Version 6.0.10 - May 04, 2020

Fixes

• Filter IPv6 Candidates from iceGatherOnlyOneCandidate SDP offer

Version 6.0.9 - March 24, 2020

Fixes

• Fix handler of rtc:transfer event to refresh conversation

Version 6.0.8 - February 28, 2020

Fixes

• Fix duplicated webrtc offer sent during IP calling
• Fix Safari WebRTC dependency

Version 6.0.7 - January 16, 2020

Fixes

• Fix ANSWERED call status in IP - IP calling
• Fix docs issues

Changes

• Improve TypeScript definitions

Version 6.0.6 - November 19, 2019

Fixes

• Add iceGatherOnlyOneCandidate configuration option and use to define path in ICE gathering process

Version 6.0.5 - November 19, 2019

Fixes

• Update styling of JSDocs to Docstrap template
• Change RTCPeerConnection ICE candidates gathering process to send the Session Description Protocol (SDP) offer on first ICE candidate gathered

Version 6.0.4 - November 14, 2019

Fixes

• Remove remaining audio elements after transferring a call to a new conversation
• Update conversation.invite() to not include empty user_id or user_name fields within the requests

Version 6.0.3 - October 22, 2019

New

• Added TypeScript definition files

Changes

• Added options for customized logging levels in the console of debug, info, warn, or error.

new NexmoClient({
  debug: 'info'
})

• Moved storage of JWT token from localStorage to NexmoClient configuration object
• Removed unnecessary files from the NPM release package

Fixes

• Fixed call statuses order in case of a transfer

Version 6.0.1 - September 27, 2019

Changes

• Removed media.record() function
• Removed cache option from SDK, used for storing conversations and events
• Removed automatic syncing of all individual conversations in login, when sync is lite or full

Version 6.0.0 - September 13, 2019

Breaking Changes

• Change return value of application.getConversations() to new ConversationsPage object

// iterate through conversations
application
  .getConversations({ page_size: 20 })
  .then((conversations_page) => {
    conversations_page.items.forEach(conversation => {
      render(conversation);
    })
  });

• Change return value of conversation.getEvents() to new EventsPage object

// iterate through events
conversation
  .getEvents({ event_type: `member:*` })
  .then((events_page) => {
    events_page.items.forEach(event => {
      render(event);
    })
  });

• Rename method application.callPhone to application.callServer
• Rename method application.call to application.inAppCall
• Rename method call.createPhoneCall to call.createServerCall
• Rename class Call to NXMCall
• Rename class ConversationClient to NexmoClient
• Rename class ConversationClientError to NexmoClientError
• Rename files conversationClient.js and conversationClient.min.js to nexmoClient.js and nexmoClient.min.js
• Deprecate member:call:state event (use instead member:call:status)
• Remove automatic login in case of a websocket reconnection and emit the event

New

• Send and listen for custom event types in a conversation.

//sending a custom event type to a conversation
conversation
  .sendCustomEvent({type: `my_custom_event`, body: { enabled: true }})
  .then((custom_event) => {
    console.log(event.body);
  });

//listening for a custom event type
conversation.on(`my_custom_event`, (from, event) => {
  console.log(event.body);
});

• Add new PageConfig class for configuring settings for paginated requests
• Add new Page class to wrap results of paginated requests
• Add setup of default pagination configuration for conversations and events in ConversationClient initialization
• Add wild card supported for filtering by event types using :* (for example event_type: member:*)

new NexmoClient({
  conversations_page_config: {
    page_size: 25,
    order: 'asc'
    cursor: 'abc'
  },
  events_page_config: {
    page_size: 50,
    event_type: `member:*`
  }
})

• Add new ConversationsPage and EventsPage which extend Page class to wrap results of paginated requests for conversations and events
• Add getNext() and getPrev() methods to ConversationsPage and EventsPage objects to fetch previous and next pages of conversations and events
• Add conversations_page_last parameter to application object and events_page_last parameter to conversation object for reference to last page retrieved

application.conversations_page_last
  .getNext((conversations_page) => {
    conversations_page.items.forEach(conversation => {
      render(conversation)
    })
  })

conversation.events_page_last
  .getPrev((events_page) => {
    events_page.items.forEach(event => {
      render(event)
    })
  })

• Add the ability to make an IP-IP call through callServer function

// IP-IP call scenario
application
  .callServer('username', 'app')
  .then((nxmCall) => {
    // console.log(nxmCall);
  });

// IP-PSTN call scenario
application
  .callServer('07400000000')
  .then((nxmCall) => {
    // console.log(nxmCall);
  });

Changes

• Update reason object to receive reason.reason_text and reason.reason_code fields

Internal changes

• Rename Event class to NXMEvent
• Update CAPI requests to REST calls for these events
  • event:delivered
  • text:delivered
  • image:delivered
  • event:seen
  • text:seen
  • image:seen
  • conversation:events
  • audio:play
  • conversation:delete
  • conversation:invite
  • text
  • text:typing:on
  • text:typing:off
  • new:conversation
  • conversation:get
  • user:conversations
  • user:get
  • conversation:join
  • audio:say
  • audio:earmuff:on
  • audio:earmuff:off
  • audio:dtmf
  • audio:record
  • audio:play
  • conversation:member:delete
  • event:delete
  • audio:ringing:start
  • audio:ringing:stop
  • audio:mute:on
  • audio:mute:off
  • image
  • rtc:new
  • rtc:answer
  • rtc:terminate
  • knocking:new
  • knocking:delete

Version 5.3.4 - July 18, 2019

Fixes

• Custom SDK config object does a deep merge with default config object

Version 5.3.3 - June 29, 2019

Fixes

• Change digits to digit in the sendDTMF() request method payload.
• Stream is not being terminated on a call transfer.
• member:call is not being emitted if media.audio_settings.enabled is false or doesn't exist.

New

• Set member.callStatus to started when initializing an IP - IP call.
• Set member.callStatus to ringing when enabling the ringing with media.startRinging().

Internal changes

• Move stream cleanup from member:left to rtc:hangup in Media module.

Version 5.2.1 - June 12, 2019

New

• Add the new nexmoGetRequest utility method to make a GET network request directly to CS:

/**
 * Perform a GET network request directly to CS
 *
 * @param {string} url the request url to CS
 * @param {string} data_type the type of data expected back from the request (events, conversations, users)
 * @param {object} [params] network request params
 * @param {string} [params.cursor] cursor parameter to access the next or previous page of a data set
 * @param {number} [params.page_size] the number of resources returned in a single request list
 * @param {string} [params.order] 'asc' or 'desc' ordering of resources (usually based on creation time)
 * @param {string} [params.event_type] the type of event used to filter event requests ('member:joined', 'audio:dtmf', etc)
 *
 * @returns {Promise<XMLHttpRequest.response>} the XMLHttpRequest.response
 * @static
 * @example <caption>Sending a nexmo GET request</caption>
 */
  nexmoGetRequest(url, data_type, params).then((response) => {
    response.body: {},
    response.cursor: {
        prev: '',
        next: '',
        self: ''
    },
    response.page_size: 10
 });

• Support reason for member:delete, conversation.leave, member.kick, call.hangup and call.reject.
• Listen for the member:left event with reason:

//listening for member:left with reason
conversation.on('member:left', (member, event) => {
  console.log(event.body.reason);
});

/**
* Reason object format
*
* @param {object} [reason] the reason for kicking out a member
* @param {string} [reason.code] the code of the reason
* @param {string} [reason.text] the description of the reason
*/

• Add callStatus field in the Member object, defining the status of a call.
• Emit member:call:status event each time the member.callStatus changes:

conversation.on("member:call:status", (member) => {
   console.log(member.callStatus);
});

Version 5.2.0 - May 30, 2019

New

• Add the call instance in application.calls map in createCall() function (IP -IP call)

• Update caller parameter in call object in a PSTN - IP call from unknown to channel.from.number or channel.from.uri if exists

• Emit the new leg:status:update event each time a member leg status change

/**
  * Conversation listening for leg:status:update events.
  *
  * @event Conversation#leg:status:update
  *
  * @property {Member} member - the member whose leg status changed
  * @property {Event} event - leg:status:update event
  * @param {string} event.cid - the conversation id
  * @param {string} event.body.leg_id - the conversation leg id
  * @param {string} event.body.type - the conversation leg type (phone or app)
  * @param {string} event.body.status - the conversation member leg status
  * @param {Array} event.body.statusHistory - array of previous leg statuses
*/
conversation.on("leg:status:update", (member, event) {
  console.log(member, event);
});

• Add the the channel.legs field in member events offered by CS

conversation.on(<member_event>, (member, event) {
  console.log(event);
  // member_id: <member_id>,
  // conversation_id: <conversation_id>,
  // ...
  // channel: {
  //  to: {
  //    type: app
  //  },
  //  type: app,
  //  leg_ids: [<leg_id>]
  //  legs : [{ leg_id: <leg_id>, status: <leg_status>}],
  //  leg_settings: {},
  // },
  // state: <state>,
  // leg_ids: []
});

---

Version 5.1.0 - May 29, 2019

New

• Send DTMF event to a conversation

  * Send DTMF in a conversation
  *
  * @param {string} digits - the DTMF digit(s) to send
  * @returns {Promise<Event>}

 conversation.media.sendDTMF('digits')

• Emit new event audio:dtmf

conversation.on("audio:dtmf",(from, event)=>{
  event.digit // the dtmf digit(s) received
  event.from //id of the user who sent the dtmf
  event.timestamp //timestamp of the event
  event.cid // conversation id the event was sent to
  event.body // additional context about the dtmf
});

• Set customized audio constraints for IP calls when enabling audio

 conversation.media.enable({
    'audioConstraints': audioConstraints
 })

  * Replaces the stream's audio tracks currently being used as the sender's sources with a new one with new audio constraints
  * @param {object} constraints - audio constraints
  * @returns {Promise<MediaStream>} - Returns the new stream with the updated audio constraints.
  * @example
  * conversation.media.updateAudioConstraints({'autoGainControl': true})
  **/

• Update audio constraints for existing audio tracks

  conversation.media.updateAudioConstraints(audioConstraints)
 })

Fixes

• Remove 'this' passed to cache worker event handler

Internal breaking changes

• Change the media audio parameter from media.audio to media.audio_settings in inviteWithAudio function

---

Version 5.0.3 - May 23, 2019

Changes

• Change default behavior of autoPlayAudio in media.enable() from false to true
• Pass an autoPlayAudio parameter to call.createCall() and call.answer() functions (default is true)

---

Version 5.0.2 - May 30, 2019

New

• Delete the image files before sending the image:delete request
• Attach of audio stream can now be chosen if it will be automatically on or off through media.enable()

media.enable({
  autoPlayAudio: true | false
})

Changes (internally)

• Combine the network GET, POST and DELETE requests in one generic function

---

Version 5.0.1 - April 30, 2019

Fixes

• Clean up user's media before leaving from an ongoing conversation

Breaking changes

• Change application.conversations type from Object to Map

---

Version 4.1.0 - April 26, 2019

Fixes

• Fixed the bug where the audio stream resolved in media.enable() is causing echo and was not the remote stream
• Resolve the remote stream pc.ontrack() and not the localStream from getUserMedia

Changes

• Rename localStream to stream in media.rtcObjects object.

---

Version 4.0.2 - April 17, 2019

Changes

• Removed media.rtcNewPromises

New

• Internal lib dependencies update
• Added support for Bugsnag error monitoring and reporting tool

 * @class ConversationClient
 *
 * @param {object} param.log_reporter configure log reports for bugsnag tool
 * @param {Boolean} param.log_reporter.enabled=false
 * @param {string} param.log_reporter.bugsnag_key your bugsnag api key / defaults to Vonage api key

• Updated vscode settings to add empty line (if none) at end of every file upon save
• Disable the ice candidates trickling in ice connection
• Wait until most of the candidates to be gathered both for the local and remote side
• Added new private function editSDPOrder(offer, answer) in rtc_helper.js to reorder the answer SDP when it's needed
• For rtc connection fail state
  • Disable leg
  • emit new event media:connection:fail

member.on("media:connection:fail",(connection_details)=>{
  connection_details.rtc_id // my member's call id / leg id
  connection_details.remote_member_id // the id of the Member the stream belongs to
  connection_details.connection_event: // the connection fail event
  connection_details.type // the type of the connection (video or screenshare)
  connection_details.streamIndex // the streamIndex of the specific stream
});

* @event Member#media:connection:fail
*
* @property {number} payload.rtc_id the rtc_id / leg_id
* @property {string} payload.remote_member_id the id of the Member the stream belongs to
* @property {event} payload.connection_event the connection fail event

• Add new LICENCE file

Breaking changes (internally)

• Deprecating ice trickling logic with onicecandidate event handler
• Change the format of member:media event to the new one offered by CS

type: 'member:media',
  from: member.member_id,
  conversation_id: member.conversation_id,
  body: {
    media: member.media,
    channel: member.channel
  }

• Change the format of member:invited event to the new offered by CS

type: 'member:invited',
  body: {
    media: {
      audio_settings: {
        enabled: false,
        earmuffed: false,
        muted: false
      }
    }
  }

---

Version 4.0.1 - March 4, 2019

New

• Select the sync level for the login process
  • full: trigger full sync to include conversations and events
  • lite: trigger partial sync, only conversation objects (empty of events)
  • none: don't sync anything

if the Cache module is enabled the manual fetch of a conversation will store them in internal storage

usage:

  new ConverationClient({'sync':'full'});

Fixes

• rtcstats:report was duplicating instances in each call
• remove screenshare https restriction

Breaking changes (internally)

• Deprecating application.activeStream, now it's part of application.activeStreams
• Removed the restriction to allow calling media.enable() while a stream is active

---

Version 4.0.0 - February 1, 2019

Breaking Changes

• rename SDK stitch to client
• listening for media:stream:* now gives streamIndex instead of index for consistency with the internal rtcObjects

 * @event Member#media:stream:on
 *
 * @property {number} payload.streamIndex the index number of this stream
 * @property {number} [payload.rtc_id] the rtc_id / leg_id
 * @property {string} [payload.remote_member_id] the id of the Member the stream belongs to
 * @property {string} [payload.name] the stream's display name
 * @property {MediaStream} payload.stream the stream that is activated
 * @property {boolean} [payload.video_mute] if the video is hidden
 * @property {boolean} [payload.audio_mute] if the audio is muted

New

• Screen Share Source ID can now be specified when invoking media.enable()