Overview

jambonz is a specification for issuing call control commands via JSON payloads over HTTP connections. These messages are sent from your application in response to web callbacks from a jambonz call control server, and they provide jambonz with your instructions on how to handle a call.

When an incoming call is received by the platform, jambonz makes an HTTP request to the URL endpoint that is configured for that called number. Outbound calls that are initiated by the REST API are controlled in a similar way -- when invoking the REST API to launch a call, you provide a web callback url and in your response to the subsequent HTTP GET or POST to that url you then return a JSON payload describing the application that should govern the outbound call.

Basic JSON message structure

The JSON payload (aka the "application") that you provide in response to a callback must be an array of objects, with each object describing a task that the platform shall perform. These tasks are executed sequentially in the order they appear in the array. Each task is identified by a verb (e.g. "dial", "gather", "hangup" etc) with associated detail and these verbs are described in more detail below.

If the caller hangs up during the execution of an application for that call, the current task is allowed to complete and any remaining tasks in the application are ignored.

Through additional callbacks that may be invoked during the execution of an application (typically as a result of those verbs which have an "action" property callback), the current application may be replaced with a new JSON application document. In such a case, the new document begins executing, and any remaining tasks in the original document are discarded.

Each task object in the JSON array must include a "verb" property that describes the action to take. Any additional information that the task needs to operate are provided as properties as well, e.g.:

{
  "verb": "say",
  "text": "Hi there!  Please leave a message at the tone and we will get back to you shortly.",
  "synthesizer": {
    "vendor": "google",
    "language": "en-US",
    "gender": "FEMALE"
  }
}

Some verbs allow other verbs to be nested; e.g. "gather" can have a nested "say" command in order to play a prompt and collect a response in one command:

{
  "verb": "gather",
  "actionHook": "/gatherCardNumber",
  "input": ["speech", "dtmf"],
  "timeout": 16,
  "numDigits": 6,
  "recognizer": {
    "vendor": "google",
    "language": "en-US"
  },
  "say": {
    "text": "Please say or enter your six digit card number now",
    "synthesizer": {
      "vendor": "google",
      "language": "en-US",
      "gender": "FEMALE"
    }
  }
}

Altogether then, a simple example application which provides the basics of a voicemail application with transcription would look like this:

[
  {
    "verb": "say",
    "text": "Hi there!  Please leave a message at the tone and we will get back to you shortly.  Thanks, and have a great day!"
  },
  {
    "verb": "listen",
    "actionHook": "http://example.com/voicemail",
    "url": "http://ws.example.com",
    "finishOnKey": "#",
    "metadata": {
      "topic": "voicemail"
    },
    "mixType": "mono",
    "playBeep": true,
    "timeout": 20,
    "transcribe": {
      "transcriptionHook": "/transcription"
    }
  },
  {
    "verb": "say",
    "text": "Thanks for your message.  We'll get back to you"
  }
]

Hey, did you see what we did there? It's a voicemail application, but where is the recording url? How do you retrieve the recording after the call?

There is none, and you don't.

Let's rethink this:

Instead of making a recording -- which exposes your customer's PII since we now have to store sensitive data at rest in the platform -- how's about we instead send you a real-time audio stream over a secure websocket connection while the call is proceeding. Annotate it with any metadata you need for tracking on your end, and we'll send that along as well.

Thus, you get the audio in real-time and we don't ever store your customer's sensitive data at rest. Bam. Done.

HTTP connection details

Each HTTP request that jambonz makes to one of your callbacks will include (at least) the following information either as query arguments (in a GET request) or in the body of the response as a JSON payload (in a POST request):

Additionally, the request MAY include

And the initial webhook for a new incoming call will have:

Finally, if you specify to use a POST method for the initial webhook for an incoming call, the JSON payload in that POST will also contain the entire incoming SIP INVITE request details in a 'sip' property (this is not provided if a GET request is used). This can be useful if you need a detailed look at all of the SIP headers or the Session Description Protocol being offered.

Note also that the information that jambonz sends you with each HTTP request can be augmented by your application by using the tag verb.

You may optionally use HTTP Basic Authentication to protect your endpoints.

call status value description
trying a new incoming call has arrived or an outbound call has just been sent
ringing a 180 Ringing response has been sent or received
early-media an early media connection has been established prior to answering the call (183 Session Progress)
in-progress call has been answered
completed an answered call has ended
failed a call attempt failed
busy a call attempt failed because the called party returned a busy status
no-answer a call attempt failed because it was not answered in time

Refer to the Example messages section to see further details.

Initial state of incoming calls

When the jambonz platform receives a new incoming call, it responds 100 Trying to the INVITE but does not automatically answer the call. It is up to your application to decide how to finally respond to the INVITE. You have some choices here.

Your application can:

The last is interesting and worthy of further comment. The intent is to let you play audio to callers without necessarily answering the call. You signal this by including an "earlyMedia" property with a value of true in the application. When receiving this, the jambonz core will create an early media connection (183 Session Progress) if possible, as shown in the example below.

Note: an early media connection will not be possible if the call has already been answered by an earlier verb in the application. In such a scenario, the earlyMedia property is ignored.

[
  {
    "verb": "say",
    "earlyMedia": true,
    "text": "Please call back later, we are currently at lunch"
    "synthesizer": {
      "vendor": "aws",
      "language": "en-US",
      "voice": "Amy"
    },
    {
      "verb": "sip:decline",
      "status": 480,
      "headers": {
        "Retry-After": 1800
      }
    }
  }
]

The say, play, gather, listen, and transcribe verbs all support the "earlyMedia" property.

The dial verb supports a similar feature of not answering the inbound call unless/until the dialed call is answered via the "answerOnBridge" property.

Speech integration

The platform makes use of text-to-speech as well as real-time speech recognition. Both google and AWS/Polly are supported for text to speech. Currently only google is supported for speech to text.

Synthesized audio is cached for up to 24 hours, so that if the same {text, language, voice} combination is requested more than once in that period it will be served from cache, reducing speech synthesis costs.

A JSON service key file containing GCP credentials for cloud speech services must be downloaded and installed on the jambonz feature servers to enable tts and speech recognition for google. For AWS/Polly, the environment variables AWS_ACCESS_KEY_ID, AWS_SECRET_ACCESS_KEY, and AWS_REGION must be provided in the server environments where the jambonz-feature-server application is running.

As part of the definition of an application, you can set defaults for the language and voice to use for speech synthesis as well as the language to use for speech recognition. These can then be overridden by verbs in the application, by using the 'synthesizer' and 'recognizer' properties./

Webhooks

Many of the verbs specify a webhook that will be called when the verb completes, or has some information to deliver to your application. These verbs contain a property that allow you to configure that webhook. By convention, the property name will always end in "Hook"; e.g "actionHook", "dtmfHook", and so on.

You can either specify the webhook as a simple string specifying a url:

"actionHook": "https://my.appserver.com/results"

or a relative url

"actionHook": "/results"

In the latter case, the base url of the application will be applied.

Alternatively, you can provide an object containing a url (required) and optional method and basic authentication parameters, e.g.:

"actionHook": {
  "url": "https://my.appserver.com/results",
  "method": "GET",
  "username": "foo",
  "password": "bar"
}

In the verb descriptions below, whenever we indicate a property is a webhook we are referring to this syntax.

Supported Verbs

Each of the supported verbs are described below.

conference

The conference verb places a call into a conference.

  {
    "verb": "conference",
    "name": "test",
    "beep": true,
    "startConferenceOnEnter": false,
    "waitHook": "/confWait",
    "enterHook": "/confEnter"   
  },

You can use the following attributes in the conference command:

option description required
actionHook A webhook to call when the conference ends no
beep if true, play a beep tone to the conference when caller enters (default: false) no
endConferenceOnExit if true, end the conference when this caller hangs up (default: false) no
enterHook A webhook to retrieve something to play or say to the caller just before they are put into a conference after waiting for it to start no
maxParticipants maximum number of participants that will be allowed in the conference no
name name of the conference yes
startConferenceOnEnter if true, start the conference only when this caller enters (default: true) no
statusHook A webhook to call with conference status events no
statusEvents An array of events for which the statusHook should be called to. See below for details. no
waitHook A webhook to retrieve commands to play or say while the caller is waiting for the conference to start no

Note: A conference bridge belongs to an Account (the Account that is associated with the Application that created the conference), and only calls generated from Applications under that Account can join that conference. The conference name property should be unique within an Account for different conference bridges; however, the same name can be used by different Accounts and each will refer to different conference bridges on the media servers.

Conference events:

Conference status webhooks will contain the following additional parameters:

dequeue

The dequeue verb removes the a call from the front of a specified queue and bridges that call to the current caller.

{
  "verb": "dequeue",
  "name": "support",
  "beep": true,
  "timeout": 60
}

You can use the following options in the dequeue command:

option description required
name name of the queue yes
actionHook A webhook invoke when call ends. If no webhook is provided, execution will continue with the next verb in the current application.
See below for specified request parameters.		 no
beep if true, play a beep tone to this caller only just prior to connecting the queued call; this provides an auditory cue that the call is now connected no
confirmHook A webhook for an application to run on the callee's end before the call is bridged. This will allow the application to play an informative message to a caller as they leave the queue (e.g. "your call may be recorded") no
timeout number of seconds to wait on an empty queue before returning (default: wait forever) no

The actionHook webhook will contain the following additional parameters:

dial

The dial verb is used to create a new call by dialing out to a number, a registered sip user, or sip endpoint. 

{
  "verb": "dial",
  "actionHook": "/outdial",
  "callerId": "+16173331212",
  "answerOnBridge": true,
  "dtmfCapture": ["*2", "*3"],
  "dtmfHook": {
    "url": "/dtmf",
    "method": "GET"
  },
  "target": [
    {
      "type": "phone",
      "number": "+15083084809"
    },
    {
      "type": "sip",
      "sipUri": "sip:1617333456@sip.trunk1.com",
      "auth": {
        "user": "foo",
        "password": "bar"
      }
    },
    {
      "type": "user",
      "name": "spike@sip.example.com"
    }
  ]
}

As the example above illustrates, when you execute the 'dial' command you are making one or more outbound call attempts in an effort to create one new call, which can be bridged to a parent call. The target property specifies an array of call destinations (aka endpoints) that will be attempted simultaneously.

If multiple endpoints are specified in the target array, all targets are outdialed at the same time (e.g., "simring", or "blast outdial" as some folks call it) and the call will be connected to the first endpoint that answers the call and, optionally, completes a call screening application as specified in the url property.

There are several types of endpoints:

You can use the following attributes in the dial command:

option description required
actionHook webhook to invoke when the call ends. no
answerOnBridge If set to true, the inbound call will ring until the number that was dialed answers the call, and at that point a 200 OK will be sent on the inbound leg. If false, the inbound call will be answered immediately as the outbound call is placed.
Defaults to false.		 no
callerId The inbound caller's phone number, which is displayed to the number that was dialed. The caller ID must be a valid E.164 number.
Defaults to caller id on inbound call.		 no
confirmHook webhook for an application to run on the callee's end after the dialed number answers but before the call is connected. This allows the caller to provide information to the dialed number, giving them the opportunity to decline the call, before they answer the call. Note that if you want to run different applications on specific destinations, you can specify the 'url' property on the nested target object. no
dialMusic url that specifies a .wav or .mp3 audio file of custom audio or ringback to play to the caller while the outbound call is ringing. no
dtmfCapture an array of strings that represent dtmf sequence which, when detected, will trigger a mid-call notification to the application via the configured dtmfHook no
dtmfHook a webhook to call when a dtmfCapture entry is matched. This is a notification only -- no response is expected, and any desired actions must be carried out via the REST updateCall API. no
headers an object containing arbitrary sip headers to apply to the outbound call attempt(s) no
listen a nested listen action, which will cause audio from the call to be streamed to a remote server over a websocket connection no
target array of to 10 destinations to simultaneously dial. The first person (or entity) to answer the call will be connected to the caller and the rest of the called numbers will be hung up. yes
timeLimit max length of call in seconds no
timeout ring no answer timeout, in seconds.
Defaults to 60.		 no
transcribe a nested transcribe action, which will cause the call to be transcribed no
target types

PSTN number

option description required
type must be "phone" yes
confirmHook A webhook for an application to run on the callee's end after the dialed number answers but before the call is connected. This will override the confirmHook property set on the parent dial verb, if any. no
number a telephone numnber in E.164 number yes

sip endpoint

option description required
type must be "sip" yes
confirmHook A webhook for an application to run on the callee's end after the dialed number answers but before the call is connected. This will override the confirmHook property set on the parent dial verb, if any. no
sipUri sip uri to send call to yes
auth authentication credentials no
auth.user sip username no
auth.password sip password no

Using this approach, it is possible to send calls out a sip trunk. If the sip trunking provider enforces username/password authentication, supply the credentials in the auth property.

a registered webrtc or sip user

option description required
type must be "user" yes
confirmHook A webhook for an application to run on the callee's end after the dialed number answers but before the call is connected. This will override the confirmHook property set on the parent dial verb, if any. no
name registered sip user, including domain (e.g. "joeb@sip.jambonz.org") yes

The confirmHook property that can be optionally specified as part of the target types (with the exception of the park type) is a web callback that will be invoked when the outdial call is answered. That callback should return an application that will run on the outbound call before bridging it to the inbound call. If the application completes with the outbound call still in a stable/connected state, then the two calls will be bridged together.

This allows you to easily implement call screening applications (e.g. "You have a call from so-and-so. Press 1 to decline").

enqueue

The enqueue command is used to place a caller in a queue.

{
    "verb": "enqueue",
    "name": "support",
    "actionHook": "/queue-action",
    "waitHook": "/queue-wait"
}

You can use the following options in the enqueue command:

option description required
name name of the queue yes
actionHook A webhook invoke when operation completes.
If a call is dequeued through the leave verb, the webook is immediately invoked.
If the call has been bridged to another party via the dequeue verb, then the webhook is invoked after both parties have disconnected.
If no webhook is provided, execution will continue with the next verb in the current application.
See below for specified request parameters.		 no
waitHook A webhook to invoke while the caller is in queue. The only allowed verbs in the application returned from this webhook are say, play, pause, and leave,
See below for additional request parameters		 no

The actionHook webhook will contain the following additional parameters:

The waitHook webhook will contain the following additional parameters:

gather

The gather command is used to collect dtmf or speech input.

{
  "verb": "gather",
  "actionHook": "http://example.com/collect",
  "input": ["digits", "speech"],
  "finishOnKey": "#",
  "numDigits": 5,
  "timeout": 8,
  "recognizer": {
    "vendor": "google",
    "language": "en-US"
  },
  "say": {
    "text": "To speak to Sales press 1.  To speak to customer support press 2.",
    "synthesizer": {
      "vendor": "google",
      "language": "en-US"
    }
  }
}

You can use the following options in the gather command:

option description required
actionHook webhook POST to invoke with the collected digits or speech. The payload will include a 'speech' or 'dtmf' property along with the standard attributes. See below for more detail. yes
finishOnKey dmtf key that signals the end of input no
input array, specifying allowed types of input: ['digits'], ['speech'], or ['digits', 'speech']. Default: ['digits'] no
numDigits number of dtmf digits expected to gather no
partialResultHook webhook to send interim transcription results to. Partial transcriptions are only generated if this property is set. no
play nested play command that can be used to prompt the user no
recognizer.hints array of words or phrases to assist speech detection no
recognizer.language language code to use for speech detection. Defaults to the application level setting, or 'en-US' if not set no
recognizer.profanityFilter if true, filter profanity from speech transcription. Default: no no
recognizer.vendor speech vendor to use (currently only google supported) no
say nested say command that can be used to prompt the user no
timeout The number of seconds of silence or inaction that denote the end of caller input. The timeout timer will begin after any nested play or say command completes. Defaults to 5 no

In the case of speech input, the actionHook payload will include a speech object with the response from google speech:

"speech": {
            "stability": 0,
            "is_final": true,
            "alternatives": [{
                "confidence": 0.858155,
                "transcript": "sales please"
            }]
        }

In the case of digits input, the payload will simple include a digits property indicating the dtmf keys pressed:

"digits": "0276"

Note: an HTTP POST will be used for both the action and the partialResultCallback since the body may need to contain nested JSON objects for speech details.

Note: the partialResultCallback web callback should not return content; any returned content will be discarded.

hangup

The hangup command terminates the call and ends the application.

{
  "verb": "hangup",
  "headers": {
    "X-Reason" : "maximum call duration exceeded"
  }
}

You can use the following options in the hangup action:

option description required
headers an object containing SIP headers to include in the BYE request no

leave

The leave verb transfers a call out of a queue. The call then returns to the flow of execution following the enqueue verb that parked the call, or the document returned by that verbs actionHook property, if provided.

{
  "verb": "leave"
}

There are no options for the leave verb.

listen

jambonz does not have a 'record' verb. This is by design, for data privacy reasons.

Recordings can contain sensitive and confidential information, and such data is never stored at rest in the jambonz core.

Instead, jambonz provides the listen verb, where an audio stream(s) can be forked and sent in real-time to a customer application for processing.

The listen verb can also be nested in a dial verb, which allows the audio for a call between two parties to be sent to a remote websocket server.

To utilize the listen verb, the customer must implement a websocket server to receive and process the audio. The endpoint should be prepared to accept websocket connections with a subprotocol name of 'audio.jambonz.org'.

The listen verb includes a url property which is the url of the remote websocket server to send the audio to. The url may be an absolute or relative URL. HTTP Basic Authentication can optionally be used to protect the websocket endpoint by using the wsAuth property.

The format of the audio data sent over the websocket is 16-bit PCM encoding, with a user-specified sample rate. The audio is sent in binary frames over the websocket connection.

Additionally, one text frame is sent immediately after the websocket connection is established. This text frame contains a JSON string with all of the call attributes normally sent on an HTTP request (e.g. callSid, etc), plus sampleRate and mixType properties describing the audio sample rate and stream(s). Additional metadata can also be added to this payload using the metadata property as described in the table below. Once the intial text frame containing the metadata has been sent, the remote side should expect to receive only binary frames, containing audio. The remote side is not expected to send any data back over the websocket.

{
  "verb": "listen",
  "url": "wss://myrecorder.example.com/calls/271314e6-b463-4980-b007-80defc181058:4433",
  "mixType" : "stereo"
}

You can use the following options in the listen action:

option description required
actionHook webhook to invoke when listen operation ends. The information will include the duration of the audio stream, and also a 'digits' property if the recording was terminated by a dtmf key. yes
finishOnKey The set of digits that can end the listen action no
maxLength the maximum length of the listened audio stream, in secs no
metadata arbitrary data to add to the JSON payload sent to the remote server when websocket connection is first connected no
mixType "mono" (send single channel), "stereo" (send dual channel of both calls in a bridge), or "mixed" (send audio from both calls in a bridge in a single mixed audio stream) Default: mono no
playBeep true, false whether to play a beep at the start of the listen operation. Default: false no
sampleRate sample rate of audio to send (allowable values: 8000, 16000, 24000, 48000, or 64000). Default: 8000 no
timeout the number of seconds of silence that terminates the listen operation. no
transcribe a nested transcribe verb no
url url of remote server to connect to yes
wsAuth.username HTTP basic auth username to use on websocket connection no
wsAuth.password HTTP basic auth password to use on websocket connection no

pause

The pause command waits silently for a specified number of seconds.

{
  "verb": "pause",
  "length": 3
}

You can use the following options in the pause action:

option description required
length number of seconds to wait before continuing the app yes

play

The play command is used to stream recorded audio to a call.

{
  "verb": "play",
  "url": "https://example.com/example.mp3"
}

You can use the following options in the play action:

option description required
url a single url or array of urls (will play in sequence) to a wav or mp3 file yes
loop number of times to play the url(s) no (default: 1)
earlyMedia if true and the call has not yet been answered, play the audio without answering call. Defaults to false no

redirect

The redirect action is used to transfer control to another JSON document taht is retrieved from the specified url. All actions after redirect are unreachable and ignored.

{
  "verb": "redirect",
  "actionHook": "/connectToSales",
}

You can use the following options in the redirect action:

option description required
actionHook URL of webhook to retrieve new application from. yes

say

The say command is used to send synthesized speech to the remote party. The text provided may be either plain text or may use SSML tags. 

{
  "verb": "say",
  "text": "hi there!",
  "synthesizer" : {
    "vendor": "google",
    "language": "en-US"
  }
}

You can use the following options in the say action:

option description required
text text to speak; may contain SSML tags yes
synthesizer.vendor speech vendor to use: google or aws (polly is also an alias for aws) no
synthesizer.language language code to use. yes
synthesizer.gender (google only) MALE, FEMALE, or NEUTRAL. no
synthesizer.voice voice to use. Note that the voice list differs whether you are using aws or google. Defaults to application setting, if provided. no
loop the number of times a text is to be repeated; 0 means repeat forever. Defaults to 1. no
earlyMedia if true and the call has not yet been answered, play the audio without answering call. Defaults to false no

sip:decline

The sip:decline action is used to reject an incoming call with a specific status and, optionally, a reason and SIP headers to include on the response.

This action must be the first and only action returned in the JSON payload for an incoming call.

The sip:decline action is a non-blocking action and the session ends immediately after the action is executed.

{
  "verb": "sip:decline",
  "status": 480,
  "reason": "Gone Fishing",
  "headers" : {
    "Retry-After": 1800
  }
}

You can use the following options in the sip:decline action:

option description required
status a valid SIP status code in the range 4XX - 6XX yes
reason a brief description no (default: the well-known SIP reasons associated with the specified status code
headers SIP headers to include in the response no

tag

The tag verb is used to add properties to the standard call attributes that jambonz includes on every action or call status HTTP POST request.

Note: because of the possible richness of the data, only subsequent POST requests will include this data. It will not be included in HTTP GET requests.

The purpose is to simplify applications by eliminating the need to store state information if it can simply be echoed back to the application on each HTTP request for the call.

For example, consider an application that wishes to apply some privacy settings on outdials based on attributes in the initial incoming call. The application could parse information from the SIP INVITE provided in the web callback when the call arrives, and rather than having to store that information for later use it could simply use the 'tag' verb to associate that information with the call. Later, when an action or call status triggers the need for the application to outdial it can simply access the information from the HTTP POST body, rather than having to retrieve it from the cache of some sort.

Note that every time the tag verb is used, the collection of customer data is completely replaced with the new data provided. This information will be provided back in all action or status notifications if POST method is used. It will appear in property named 'customerData' in the JSON payload. 

{
  "verb": "tag",
  "data" {
        "foo": "bar",
        "counter": 100,
        "list": [1, 2, "three"]
    }
}

After the above 'tag' verb has executed, web callbacks using POST would have a payload similar to this:

{
    "callSid": "df09e8d4-7ffd-492b-94d9-51a60318552c",
    "direction": "inbound",
    "from": "+15083084809",
    "to": "+15083728299",
    "callId": "f0414693-bdb6-1238-6185-06d91d68c9b0",
    "sipStatus": 200,
    "callStatus": "in-progress",
    "callerId": "f0414693-bdb6-1238-6185-06d91d68c9b0",
    "accountSid": "fef61e75-cec3-496c-a7bc-8368e4d02a04",
    "applicationSid": "0e0681b0-d49f-4fb8-b973-b5a3c6758de1",
    "originatingSipIp": "54.172.60.1:5060",
    "originatingSipTrunkName": "twilio",
    "customerData": {
        "foo": "bar",
        "counter": 100,
        "list": [1, 2, "three"]
    }
}

You can use the following options in the tag command:

option description required
data a JSON object containing values to be saved and included in future action or call status notifications (HTTP POST only) for this call yes

transcribe

The transcribe verb is used to send real time transcriptions of speech to a web callback.

The transcribe command is only allowed as a nested verb within a dial or listen verb. Using transcribe in a dial command allows a long-running transcription of a phone call to be made, while nesting within a listen verb allows transcriptions of recorded messages (e.g. voicemail).

{
  "verb": "transcribe",
  "transcriptionHook": "http://example.com/transcribe",
  "recognizer": {
    "vendor": "google",
    "language" : "en-US",
    "interim": true
  }
}

You can use the following options in the transcribe command:

option description required
recognizer.dualChannel if true, transcribe the parent call as well as the child call no
recognizer.interim if true interim transcriptions are sent no (default: false)
recognizer.language language to use for speech transcription yes
recognizer.profanityFilter if true, filter profanity from speech transcription. Default: no no
recognizer.vendor speech vendor to use (currently only google supported) no
transcriptionHook webhook to call when a transcription is received. Due to the richness of information in the transcription an HTTP POST will always be sent. yes

Note: the dualChannel property is not currently implemented.

Example messages

An example JSON payload for a webhook for an incoming call using a POST method. There's a lot of detail here, because when you specify to receive a POST you are getting the full SIP INVITE.

{
    "direction": "inbound",
    "callSid": "1fe62f7c-ebb9-4b96-b75b-7d04ff2b195d",
    "accountSid": "fef61e75-cec3-496c-a7bc-8368e4d02a04",
    "applicationSid": "0e0681b0-d49f-4fb8-b973-b5a3c6758de1",
    "from": "+15083084809",
    "to": "+15083728299",
    "callerName": "+15083084809",
    "callId": "252a93d3-bdb2-1238-6185-06d91d68c9b0",
    "sipStatus": 100,
    "callStatus": "trying",
    "originatingSipIp": "54.172.60.2:5060",
    "originatingSipTrunkName": "twilio",
    "sip": {
        "headers": {
            "via": "SIP/2.0/UDP 3.10.235.99;rport=5060;branch=z9hG4bKgeBy6Fg863Z8N;received=172.31.3.33",
            "max-forwards": "70",
            "from": "<sip:+15083084809@3.10.235.99:5060>;tag=vQXQ3g5papXpF",
            "to": "<sip:+15083728299@172.31.3.33:5070>",
            "call-id": "252a93d3-bdb2-1238-6185-06d91d68c9b0",
            "cseq": "15623387 INVITE",
            "contact": "<sip:+15083084809@3.10.235.99:5060>",
            "user-agent": "Twilio Gateway",
            "allow": "INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS",
            "content-type": "application/sdp",
            "content-length": "264",
            "X-CID": "f9221ea5e66a1d1f10a0b556933dc0c2@0.0.0.0",
            "X-Forwarded-For": "54.172.60.2:5060",
            "X-Originating-Carrier": "twilio",
            "Diversion": "<sip:+15083728299@public-vip.us1.twilio.com>;reason=unconditional"
        },
        "body": "v=0\r\no=root 1999455157 1999455157 IN IP4 3.10.235.99\r\ns=Twilio Media Gateway\r\nc=IN IP4 3.10.235.99\r\nt=0 0\r\nm=audio 49764 RTP/AVP 0 101\r\na=maxptime:150\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=sendrecv\r\na=rtcp:49765\r\na=ptime:20\r\n",
        "payload": [{
            "type": "application/sdp",
            "content": "v=0\r\no=root 1999455157 1999455157 IN IP4 3.10.235.99\r\ns=Twilio Media Gateway\r\nc=IN IP4 3.10.235.99\r\nt=0 0\r\nm=audio 49764 RTP/AVP 0 101\r\na=maxptime:150\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=sendrecv\r\na=rtcp:49765\r\na=ptime:20\r\n"
        }],
        "method": "INVITE",
        "version": "2.0",
        "uri": "sip:+15083728299@172.31.3.33:5070",
        "raw": "INVITE sip:+15083728299@172.31.3.33:5070 SIP/2.0\r\nVia: SIP/2.0/UDP 3.10.235.99;rport=5060;branch=z9hG4bKgeBy6Fg863Z8N;received=172.31.3.33\r\nMax-Forwards: 70\r\nFrom: <sip:+15083084809@3.10.235.99:5060>;tag=vQXQ3g5papXpF\r\nTo: <sip:+15083728299@172.31.3.33:5070>\r\nCall-ID: 252a93d3-bdb2-1238-6185-06d91d68c9b0\r\nCSeq: 15623387 INVITE\r\nContact: <sip:+15083084809@3.10.235.99:5060>\r\nUser-Agent: Twilio Gateway\r\nAllow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS\r\nContent-Type: application/sdp\r\nContent-Length: 264\r\nX-CID: f9221ea5e66a1d1f10a0b556933dc0c2@0.0.0.0\r\nX-Forwarded-For: 54.172.60.2:5060\r\nX-Originating-Carrier: twilio\r\nDiversion: <sip:+15083728299@public-vip.us1.twilio.com>;reason=unconditional\r\nX-Twilio-AccountSid: AC58f23d38858ac262d6ee2e554b30c561\r\nX-Twilio-CallSid: CA708d85d118aacfcc794b730fa02bc40c\r\n\r\nv=0\r\no=root 1999455157 1999455157 IN IP4 3.10.235.99\r\ns=Twilio Media Gateway\r\nc=IN IP4 3.10.235.99\r\nt=0 0\r\nm=audio 49764 RTP/AVP 0 101\r\na=maxptime:150\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=sendrecv\r\na=rtcp:49765\r\na=ptime:20\r\n"
    }
}

An example JSON payload for a call status webhook for an incoming call using a POST method:

 {
    "direction": "inbound",
    "callSid": "1fe62f7c-ebb9-4b96-b75b-7d04ff2b195d",
    "accountSid": "fef61e75-cec3-496c-a7bc-8368e4d02a04",
    "applicationSid": "0e0681b0-d49f-4fb8-b973-b5a3c6758de1",
    "from": "+15083084809",
    "to": "+15083728299",
    "callerName": "+15083084809",
    "callId": "252a93d3-bdb2-1238-6185-06d91d68c9b0",
    "sipStatus": 200,
    "callStatus": "in-progress",
    "originatingSipIp": "54.172.60.2:5060",
    "originatingSipTrunkName": "twilio"
 }

An example JSON payload for a call status webhook for an outbound call using a POST method:

{
    "direction": "outbound",
    "callSid": "ddd6d4b2-ba3f-42fb-9845-8abdac047097",
    "parentCallSid": "1fe62f7c-ebb9-4b96-b75b-7d04ff2b195d",
    "accountSid": "fef61e75-cec3-496c-a7bc-8368e4d02a04",
    "applicationSid": "0e0681b0-d49f-4fb8-b973-b5a3c6758de1",
    "from": "+15083084809",
    "to": "+15084901000",
    "callerName": "+15083084809",
    "callId": "a5726393-bdaf-1238-9483-06d91d68c9b0",
    "callStatus": "in-progress",
    "sipStatus": 200
}