Session Initiation Protocol (SIP)

Updated: Nov 18, 2025 When SIP is enabled, you cannot use calling related Graph API endpoints and calling related webhooks are not sent.

Overview

Session Initiation Protocol ( SIP) is a signaling protocol used for initiating, maintaining, modifying, and terminating real-time communication sessions between two or more endpoints. WhatsApp Business Calling API supports use of SIP as the signaling protocol instead of our Graph API endpoints and Webhooks.

Before you get started

Before you get started with SIP call signaling, confirm the following: You meet overall calling pre-requisites Your app has messaging permissions for the business phone number you want to enable SIP for. Test this by sending and receiving messages using Graph API messaging endpoints, then use the same app to configure your SIP server on the business phone number for calling. Double confirm this by using health status API with PHONE_NUMBER_ID Your app mode is “Live”, not “Development”. You have a standards compliant third party SIP server that supports TLS transport and digest authentication

Signaling and media possible configurations

	Default configuration after enabling calling	SIP with WebRTC	SIP with SDES media
Signaling protocol	Graph APIs + Webhooks	SIP (needs explicit enablement)	SIP (needs explicit enablement)
Signaling transport	HTTPS	TLS	TLS
Media protocol	WebRTC (ICE + DTLS* + SRTP)	WebRTC (ICE + DTLS + SRTP)	SDES SRTP (needs explicit enablement)
Audio codec	OPUS	OPUS	OPUS

*Note: You can use SDES instead of ICE+DTLS with Graph API + Webhook signaling

Calling flows using SIP

Before you start, make sure you have enabled and configured SIP on the business phone number. Meta generates a unique SIP user password for each business phone number + app combination. You will need this information and can retrieve it by using the get Call Settings endpoint.

Security

TLS transport is mandatory for SIP. Meta will present a valid server cert with subject name that covers our SIP domain wa.meta.vc. Your SIP server should do the same as Meta ensures your cert is valid and subject name covers SIP domain you configured on the business phone number Meta does NOT support mutual TLS (aka mTLS). This means, when Meta takes the role of a TLS client, your TLS server should not request Client certificate. If you still request client cert, Meta will present a client cert but the cert subject name would refer to a random dynamic host which will not pass certificate validation. Meta adds transport=TLS to request URI as part of it’s SIP requests to partner SIP server For business initiated calls, SIP invite from your SIP server will be challenged using digest auth. See business-initiated calls for more details For user initiated calls, it is highly recommended that you challenge SIP INVITE request from Meta, to use digest auth for more security. See user-initiated calls for more details

How to test if you have a valid TLS certificate

When a WhatsApp user calls a business, a common reason for your SIP server to not receive the SIP INVITE from Meta is the certificate validation error. You can use information here to confirm valid setup. Run the command openssl s_client -quiet -verify_hostname {hostname} -connect {hostname}:{port} by properly substituting hostname and port with your values

Example of valid server cert

$ openssl s_client -quiet -verify_hostname meta-voip.example.com -connect meta-voip.example.com:5061
Connecting to 64:ff9b::68f8:b0b8
depth=2 C=US, ST=New Jersey, L=Jersey City, O=The USERTRUST Network, CN=USERTrust RSA Certification Authority
verify return:1
depth=1 C=AT, O=ZeroSSL, CN=ZeroSSL RSA Domain Secure Site CA
verify return:1
depth=0 CN=example.com
verify return:1

Example of hostname:port not listening on TLS

openssl s_client -quiet -verify_hostname lb01.voice.usw2.pure.cloud -connect lb01.voice.usw2.pure.cloud:5060
Connecting to 34.211.206.63
009F0DFB01000000:error:0A000126:SSL routines::unexpected eof while reading:ssl/record/rec_layer_s3.c:693:

Example of invalid cert

$ openssl s_client -quiet -verify_hostname meta-inb.byoc.mypurecloud.com -connect meta-inb.byoc.mypurecloud.com:5061
Connecting to 64:ff9b::3652:f1c0
depth=0 jurisdictionC=US, jurisdictionST=California, businessCategory=Private Organization, serialNumber=1515861, C=US, ST=Indiana, L=Indianapolis, O=Genesys Cloud Services, Inc., CN=voice.mypurecloud.com
verify error:num=62:hostname mismatch
verify return:1
depth=2 C=US, O=DigiCert Inc, OU=www.digicert.com, CN=DigiCert High Assurance EV Root CA
verify return:1
depth=1 C=US, O=DigiCert Inc, OU=www.digicert.com, CN=DigiCert SHA2 Extended Validation Server CA
verify return:1
depth=0 jurisdictionC=US, jurisdictionST=California, businessCategory=Private Organization, serialNumber=1515861, C=US, ST=Indiana, L=Indianapolis, O=Genesys Cloud Services, Inc., CN=voice.mypurecloud.com
verify return:1

In this case, you can alter the certificate to match your hostname or change your configured SIP server hostname to match your certificate

Business-initiated Calls

Prerequisites

You have the required call permission approval from the WhatsApp user Learn how to obtain user calling permissions Retrieve Meta generated SIP password and configure it on your SIP server, so it can respond to digest authentication challenge from Meta SIP servers

Calling flow

Send an initial SIP INVITE to our servers. Our SIP domain is wa.meta.vc. To initiate a call to WhatsApp user with phone number 11234567890, the SIP request URI should be ‘sip:[email protected];transport=tls’ This request will fail with an “SIP 407 Proxy Authentication required” message. Send a 2nd SIP INVITE with proper Authorization header as per RFC 3261. The Authorization field’s username attribute must match the from header’s user name which is the business phone number The password is generated by Meta and you can retrieve it using get Call Settings endpoint The username portion of the from header must be the fully normalized business phone number The domain name of the from header must match the SIP server you configured on the business phone number The SDP Offer you include supports ICE, DTLS-SRTP and OPUS (essentially WebRTC media) Send the SIP INVITE to the WhatsApp user number you want to call.

User-initiated Calls

Prerequisites

If you plan to use SIP Digest Auth, retrieve Meta generated SIP password and configure it on your SIP server, so it can respond to digest authentication challenge from Meta SIP servers

Calling flow

The WhatsApp user calls business phone number and is unaware of whether the business is using SIP or Graph API. In other words, the user experience is identical If the business phone number is SIP enabled, Meta will send an SIP INVITE to the SIP server configured on the business phone number You respond with SIP digest auth challenge (recommended) or SIP OK and pass in an SDP answer If you are not receiving SIP INVITE from Meta, refer to SIP specific FAQ to troubleshoot further View sample SIP requests Learn more about Session Description Protocol (SDP) View example SDP structures

Custom SIP Headers

We use following custom SIP headers specific to user-initiated calls

Header name	Metadata	Description
x-wa-meta-cta-payload	Optional; String	Present when user-initiates a call from call button that has business specified payload. Learn more
x-wa-meta-deeplink-payload	Optional; String	Present when user-initiates a call from call deeplink that has business specified payload. Learn more

Configure/Update SIP settings on business phone number

Use this endpoint to update call settings configuration for an individual business phone number.

Request syntax

POST /&lt;PHONE_NUMBER_ID&gt;/settings

Endpoint parameters

Placeholder	Description	Sample Value
`<PHONE_NUMBER_ID>` Integer	Required The business phone number for which you are updating Calling API settings. Learn more about formatting phone numbers in Cloud API	`+12784358810`

Request body

\{
  "calling": {
    ... // other calling api settings
    "sip": {
      "status": "ENABLED",
      "servers": [\
        {\
         "hostname": SIP_SERVER_HOSTNAME\
          "port": SIP_SERVER_PORT,\
          "request_uri_user_params": {\
            "KEY1": "VALUE1", // for cases like trunk groups (tgrp)\
            "KEY2": "VALUE2",\
          }\
        }\
      ]
    }
  },
  // Other non calling api feature configurations
}

Body parameters

Parameter	Description	Sample Value
`status` String	Optional Enable or disable SIP call signaling for the given business phone number. Default is `DISABLED`. When `status` is `ENABLED`, this phone number will exclusively use SIP for call signaling and will not work with Graph APIs. No webhooks are sent. When `status` is set to `DISABLED`, the SIP `servers` values are not reset. If you enable SIP on the same phone number again, the previously configured `servers` values will take effect. You can configure both status and SIP servers in the same request	`“ENABLED”` `“DISABLED”`
`servers` String	Optional The SIP server routing configuration. Each application can have only 1 SIP server configured for it. The servers is an array to be futureproof. It also makes the POST payload schema be consistent with GET payload schema because you can have multiple apps each with their own SIP server. In the GET payload, if you see multiple SIP servers, it means you’ve used the POST API with different access tokens that belong to different apps. The associated app is extracted from the access token used to make the API call. Since a business phone number may be used with multiple applications, each number can be serviced by multiple SIP servers. To delete a previously configured SIP server, pass an empty array to this field. If you still see some servers remaining after you clear, those servers may belong to different apps, so you need to use the corresponding access tokens to clear them `hostname` — ( String) Required The host name of the SIP server. Requests must use TLS. `port` — ( String) Required The port within your SIP server that will accept requests. Requests must use TLS. Default port is 5061 `request_uri_user_params` — ( String) Optional An optional field for passing any parameters you want included in the user portion of the request URI used in our SIP INVITE to your SIP server. Max key/value size is 128 characters. An example use case would be Trunk Groups ( RFC 4904) sip:[email protected]; tgrp=wacall; trunk-context=byoc.example.com This example has two user parameters for tgrp, and trunk-context.	`<br />"servers": [<br /> {<br /> "hostname": SIP_SERVER_HOSTNAME<br /> "port": SIP_SERVER_PORT,<br /> "request_uri_user_params": {<br /> "KEY1": "VALUE1",<br /> "KEY2": "VALUE2",<br /> }<br /> }<br />]<br />`

Success response

\{
  "success": true
}

Error response

View Calling API Error Codes and Troubleshooting for more information View general Cloud API Error Codes here

Get phone number calling settings (SIP)

Use this endpoint to check the configuration of your Calling API feature settings, including SIP values. This endpoint can return information for other Cloud API feature settings.

Request syntax

GET /&lt;PHONE_NUMBER_ID&gt;/settings

Endpoint parameters

Placeholder	Description	Sample Value
`<PHONE_NUMBER_ID>` Integer	Required The business phone number for which you are retrieving Calling API settings. Learn more about formatting phone numbers in Cloud API	`+12784358810`

App Permission Required

whatsapp_business_management: Advanced access is required to update use the API for end business clients

Response body

\{
  "calling": {
    "status": "ENABLED",
    "call_icon_visibility": "DEFAULT",
    "callback_permission_status": "ENABLED",
    "sip": {
      "status": "ENABLED",
      "servers": [\
        {\
          "app_id": &lt;APP_ID_THAT_CONFIGURED_THIS_SIP_SERVER&gt;,\
          "hostname": "sip.example.com"\
        }\
      ]
    }
  }
}

Include SIP User Password

Optionally, you can include SIP user credentials in your response body by adding the SIP credentials query parameter in the POST request:

GET /&lt;PHONE_NUMBER_ID&gt;/settings?include_sip_credentials=true

Where the response will look like this:

\{
  "calling": {
    "status": "ENABLED",
    "call_icon_visibility": "DEFAULT",
    "callback_permission_status": "ENABLED",
    "sip": {
      "status": "ENABLED",
      "servers": [\
        {\
          "app_id": &lt;APP_ID_THAT_CONFIGURED_THIS_SIP_SERVER&gt;,\
          "hostname": "sip.example.com",\
          "sip_user_password": "{SIP_USER_PASSWORD}"\
        }\
      ]
    }
  }
}

Error response

View Calling API Error Codes and Troubleshooting for more information View general Cloud API Error Codes here

Reset SIP password

To make Meta generate a new SIP password, you’d need to disable SIP, delete SIP server and add your SIP server back. Fetch your SIP configuration with password to view your current password for your reference Disable and delete your SIP server

curl -X POST \
https://graph.facebook.com/{VERSION}/{PHONE_NUMBER_ID}/settings \
-H 'Authorization: Bearer {TOKEN}' \
-H 'Content-Type: application/json' \
-d '
\{
  "calling": {
    "status": "DISABLED",
    "sip": {
      "status": "DISABLED",
      "servers": [],
    }
  }
}'
{"success":true}

Enable SIP and add your SIP server

curl -X POST \
https://graph.facebook.com/{VERSION}/{PHONE_NUMBER_ID}/settings \
-H 'Authorization: Bearer {TOKEN}' \
-H 'Content-Type: application/json' \
-d '
\{
  "calling": {
    "status": "ENABLED",
    "sip": {
      "status": "ENABLED",
      "servers": [{"hostname":"sip.example.com"}],
    }
  }
}'
{"success":true}

Fetch your SIP configuration with password to note the new password

Sample SIP requests

Business-initiated calls (with WebRTC media)

Initial SIP INVITE request

INVITE sip:[email protected];transport=tls SIP/2.0
Record-Route: <sip:+159.65.244.171:5061;transport=tls;lr;ftag=Kc9QZg4496maQ;nat=yes>
Via: SIP/2.0/TLS 159.65.244.171:5061;received=2803:6081:798c:93f8:5f9b:bfe8:300:0;branch=z9hG4bK0da2.36614b8977461b486ceabc004c723476.0;i=617261
Via: SIP/2.0/TLS 137.184.87.1:35181;rport=56533;received=137.184.87.1;branch=z9hG4bKQNa6meey5Dj2g
Max-Forwards: 69
From: <sip:[email protected]>;tag=Kc9QZg4496maQ
To: <sip:[email protected]>
Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba
CSeq: 96743476 INVITE
Contact: <sip:[email protected]:35181;transport=tls;swrad=137.184.87.1~56533~3>
User-Agent: SignalWire
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 600;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 2427
X-Relay-Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
Content-Type: application/sdp
Content-Length:  2427

<<SDP omitted for brevity>>

407 response from Meta

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TLS 159.65.244.171:5061;received=2803:6081:798c:93f8:5f9b:bfe8:300:0;branch=z9hG4bK0da2.36614b8977461b486ceabc004c723476.0;i=617261
Via: SIP/2.0/TLS 137.184.87.1:35181;rport=56533;received=137.184.87.1;branch=z9hG4bKQNa6meey5Dj2g
Record-Route: <sip:+159.65.244.171:5061;transport=tls;lr;ftag=Kc9QZg4496maQ;nat=yes>
Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba
From: <sip:[email protected]>;tag=Kc9QZg4496maQ
To: <sip:[email protected]>;tag=z9hG4bK0da2.36614b8977461b486ceabc004c723476.0
CSeq: 96743476 INVITE
Proxy-Authenticate: Digest realm="wa.meta.vc",nonce="419ac2415577f8e1",opaque="440badfc05072367",algorithm=MD5,qop="auth"

Second SIP INVITE send with authorization

INVITE sip:[email protected];transport=tls SIP/2.0
        Record-Route: <sip:+159.65.244.171:5061;transport=tls;lr;ftag=Kc9QZg4496maQ;nat=yes>
        Via: SIP/2.0/TLS 159.65.244.171:5061;received=2803:6081:798c:93f8:5f9b:bfe8:300:0;branch=z9hG4bK1da2.ed8900012befced853927008d619d374.0;i=617261
        Via: SIP/2.0/TLS 137.184.87.1:35181;rport=56533;received=137.184.87.1;branch=z9hG4bKry3yp9y12p8mc
        Max-Forwards: 69
        From: <sip:[email protected]>;tag=Kc9QZg4496maQ
        To: <sip:[email protected]>
        Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba
        CSeq: 96743477 INVITE
        Contact: <sip:[email protected]:35181;transport=tls;swrad=137.184.87.1~56533~3>
        User-Agent: SignalWire
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
        Supported: timer, path, replaces
        Allow-Events: talk, hold, conference, refer
        Proxy-Authorization: Digest username="17125550259", realm="wa.meta.vc", nonce="419ac2415577f8e1", uri="sip:[email protected];transport=tls", response="blah", algorithm=MD5, cnonce="/mVZtYFCEj65YQJCrBEAAg", opaque="440badfc05072367", qop=auth, nc=00000001
        Session-Expires: 600;refresher=uac
        Min-SE: 90
        Content-Type: application/sdp
        Content-Disposition: session
        Content-Length: 2427
        X-Relay-Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba
        Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
        Content-Type: application/sdp
        Content-Length:  2427
        <<SDP omitted for brevity>>

Example error response

SIP/2.0 403 SIP server wa.meta.vc from INVITE does not match any SIP server configured for phone number id {ID}
        Via: SIP/2.0/TLS [2803:6080:c954:b533:ecfb:5cec:300:0]:39459;rport=39459;received=2803:6080:c954:b533:ecfb:5cec:300:0;branch=z9hG4bKPjf9f3d0bddb3dbe0c9b1e3b486f39784a;alias
        Via: SIP/2.0/TLS 148.72.155.236:5061;rport=30498;received=2803:6080:d014:8e40:ddbb:4ed7:300:0;branch=z9hG4bKPjfd270ec8-7aaf-4a65-b290-4bef3b50b7b7;alias
        Record-Route: <sip:onevc-sip-proxy-dev.fbinfra.net:8191;transport=tls;lr>
        Record-Route: <sip:wa.meta.vc;transport=tls;lr>
        Call-ID: 91578781-44f1-4268-9a7f-d7efec1abf72
        From: <sip:[email protected]>;tag=3a63b370-a697-4a5a-82b4-e8105e23f176
        To: <sip:[email protected]>;tag=e0d30a05-657b-47ec-a668-e05ca79f9f05
        CSeq: 15659 INVITE
        Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
        X-FB-External-Domain: wa.meta.vc
        Warning: 399 wa.meta.vc "SIP server wa.meta.vc from INVITE does not match any SIP server configured for phone number id {ID}"
        Content-Length: 0
        Content-Length:  0

SIP BYE

BYE sip:[email protected];transport=tls;ob SIP/2.0
Via: SIP/2.0/TLS 137.184.4.155:5061;received=2803:6080:c074:cac:10ed:4b05:400:0;i=8d2dc2
Via: SIP/2.0/TLS 143.198.136.243:35181;rport=38087;received=143.198.136.243
Route: <sip:wa.meta.vc;transport=tls;lr>
Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Max-Forwards: 69
From: <sip:[email protected]>;tag=NcKQ6mtDKSDQB
To: "5559800000693" <sip:[email protected]>;tag=92a01092-ee78-4870-865f-bc176203a6bd
Call-ID: outgoing:wacid.HBgPMjAwNzU2OTA0ODY5OTY1FRIAEhggMjQ4QzUwOUQ1REQ0NDUwNENEQzRFMTgwRTNGQjAwNjEcGAsxMjE0NTU1MTg2ORUCAAA
CSeq: 98734935 BYE
User-Agent: SignalWire
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
X-Relay-Call-ID: b72c0c65-e319-41b3-afb7-19ebcca05d38
Content-Length:  0

SIP Invite (with SDES)

INVITE sip:[email protected];transport=tls SIP/2.0
Record-Route: <sip:54.172.60.1:5061;transport=tls;lr;r2=on>
Record-Route: <sip:54.172.60.1;lr;r2=on>
CSeq: 2 INVITE
From: "12145551869" <sip:[email protected]>;tag=28460006_c3356d0b_5cdada8c-cbf0-4369-b02d-cc97d3c36f2b
To: <sip:[email protected]>
Max-Forwards: 66
P-Asserted-Identity: <sip:[email protected]>
Min-SE: 120
Call-ID: [email protected]
Contact: "12145551869" <sip:[email protected]:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Via: SIP/2.0/TLS 54.172.60.1:5061;received=2803:6080:f934:8894:7eb5:24f9:300:0;branch=z9hG4bK1e5a.0da2ace9cc912d9e5f2595ca4acb9847.0
Via: SIP/2.0/UDP 172.25.10.217:5060;rport=5060;branch=z9hG4bK5cdada8c-cbf0-4369-b02d-cc97d3c36f2b_c3356d0b_54-457463274351249162
Supported: timer
User-Agent: Twilio Gateway
Proxy-Authorization: Digest username="12145551869", realm="wa.meta.vc", nonce="2a487cb01d4ed43b", uri="sip:[email protected];transport=tls", response="3f58df7af575b948625aeffd51bf9060", algorithm=MD5, cnonce="b338deb7f0e004e66353e26d34ad62b7", opaque="725a06fb2cd89a32", qop=auth, nc=00000002
Content-Type: application/sdp
X-Twilio-CallSid: CA93eac6be615da5e6836c7059e9555348
Content-Length: 422
Content-Type: application/sdp
Content-Length:   422

v=0
o=root 1185414872 1185414872 IN IP4 172.18.155.180
s=Twilio Media Gateway
c=IN IP4 168.86.138.232
t=0 0
m=audio 19534 RTP/SAVP 107 0 8 101
a=crypto:**************************************************************************
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

SIP OK (with SDES)

SIP/2.0 200 OK
Via: SIP/2.0/TLS 54.172.60.1:5061;received=2803:6080:f934:8894:7eb5:24f9:300:0;branch=z9hG4bK1e5a.0da2ace9cc912d9e5f2595ca4acb9847.0
Via: SIP/2.0/UDP 172.25.10.217:5060;rport=5060;branch=z9hG4bK5cdada8c-cbf0-4369-b02d-cc97d3c36f2b_c3356d0b_54-457463274351249162
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:54.172.60.1:5061;transport=tls;lr;r2=on>
Record-Route: <sip:54.172.60.1;lr;r2=on>
Call-ID: [email protected]
From: "12145551869" <sip:[email protected]>;tag=28460006_c3356d0b_5cdada8c-cbf0-4369-b02d-cc97d3c36f2b
To: <sip:[email protected]>;tag=0d185053-2615-46c7-8ff2-250bda494cf1
CSeq: 2 INVITE
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
Supported: timer
X-FB-External-Domain: wa.meta.vc
Contact: <sip:[email protected];transport=tls;ob;X-FB-Sip-Smc-Tier=collaboration.sip_gateway.sip.prod>;isfocus
Content-Type: application/sdp
Content-Length:   645

v=0
o=- 1746657286595 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 42da9643-cb50-4eca-95d3-ca41b3f1f4bb
m=audio 3480 RTP/SAVP 107 101
c=IN IP4 157.240.19.130
a=rtcp:9 IN IP4 0.0.0.0
a=mid:audio
a=sendrecv
a=msid:42da9643-cb50-4eca-95d3-ca41b3f1f4bb WhatsAppTrack1
a=rtcp-mux
a=crypto:**************************************************************************
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=maxptime:20
a=ptime:20
a=ssrc:1238967757 cname:WhatsAppAudioStream1

User-initiated calls (with WebRTC media)

SIP INVITE

INVITE sip:[email protected];transport=tls SIP/2.0
Via: SIP/2.0/TLS [2803:6080:e888:51aa:d4a4:c5e0:300:0]:33819;rport=33819;received=2803:6080:e888:51aa:d4a4:c5e0:300:0;branch=z9hG4bKPjNvs.IZBnUa1W4l8oHPpk3SUMmcx3MMcE;alias
Max-Forwards: 70
From: "12195550714" <sip:[email protected]>;tag=bbf1ad6e-79bb-4d9c-8a2c-094168a10bea
To: <sip:[email protected]>
Contact: <sip:[email protected];transport=tls;ob>;isfocus
Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCAzODg1NTE5NEU1NTBEMTc1RTFFQUY5NjNCQ0FCRkEzRhwYCzE3MDE1NTU4ODU3FQIAAA==
CSeq: 2824 INVITE
Route: <sip:onevc-sip-proxy-dev.fbinfra.net:8191;transport=tls;lr>
X-FB-External-Domain: wa.meta.vc
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
User-Agent: Facebook SipGateway
Content-Type: application/sdp
Content-Length: 1028

v=0
o=- 1741113186367 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 632a909f-1060-4369-96a4-7bd03e291ee7
a=ice-lite
m=audio 3480 UDP/TLS/RTP/SAVPF 111 126
c=IN IP4 57.144.135.35
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1775469887 1 udp 2122260223 57.144.135.35 3480 typ host generation 0 network-cost 50
a=candidate:3355715111 1 udp 2122262783 2a03:2880:f343:131:face:b00c:0:699c 3480 typ host generation 0 network-cost 50
a=ice-ufrag:RmDDkfzkwbexPfbC
a=ice-pwd:*************************
a=fingerprint:********************************************************************************************************
a=setup:actpass
a=mid:audio
a=sendrecv
a=msid:632a909f-1060-4369-96a4-7bd03e291ee7 WhatsAppTrack1
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=maxptime:20
a=ptime:20
a=ssrc:849255537 cname:WhatsAppAudioStream1

SIP BYE

BYE sip:[email protected];transport=tls;ob SIP/2.0
Via: SIP/2.0/TLS 137.184.4.155:5061;received=2803:6080:c074:cac:10ed:4b05:400:0;i=8d2dc2
Via: SIP/2.0/TLS 143.198.136.243:35181;rport=38087;received=143.198.136.243
Route: <sip:wa.meta.vc;transport=tls;lr>
Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Max-Forwards: 69
From: <sip:[email protected]>;tag=NcKQ6mtDKSDQB
To: "5559800000693" <sip:[email protected]>;tag=92a01092-ee78-4870-865f-bc176203a6bd
Call-ID: outgoing:wacid.HBgPMjAwNzU2OTA0ODY5OTY1FRIAEhggMjQ4QzUwOUQ1REQ0NDUwNENEQzRFMTgwRTNGQjAwNjEcGAsxMjE0NTU1MTg2ORUCAAA
CSeq: 98734935 BYE
User-Agent: SignalWire
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
X-Relay-Call-ID: b72c0c65-e319-41b3-afb7-19ebcca05d38
Content-Length:  0

SIP Invite (with SDES)

INVITE sip:[email protected];transport=tls SIP/2.0
            Via: SIP/2.0/TLS [2803:6080:f948:9597::]:57363;rport;branch=z9hG4bKPj3a9f2ad89e4a3df61408aa84f7d9a63e;alias
            Record-Route: <sip:wa.meta.vc;transport=tls;lr>
            Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
            Via: SIP/2.0/TLS [2803:6080:f948:9597:d33c:e00:400:0]:5061;branch=z9hG4bKPj3a9f2ad89e4a3df61408aa84f7d9a63e
            Via: SIP/2.0/TLS [2803:6080:f948:9597:1ac5:cdf8:300:0]:63057;rport=63057;received=2803:6080:f948:9597:1ac5:cdf8:300:0;branch=z9hG4bKPj-phic0sbns27DiP0OlrxRxgLtNg4mio7;alias
            Max-Forwards: 69
            From: "12195550714" <sip:[email protected]>;tag=8a0f7e65-6e9e-4801-bf92-85c3ef2485d9
            To: <sip:[email protected]>
            Contact: <sip:[email protected];transport=tls;ob>;isfocus
            Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCA4QkY1MTJCQkNFNTgxMEVFRERFRTUzNTFERkE1MDU0MhwYCzEyMTQ1NTUxODY5FQIAAA
            CSeq: 31159 INVITE
            X-FB-External-Domain: wa.meta.vc
            Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
            User-Agent: Facebook SipGateway
            Content-Type: application/sdp
            Content-Length:   645

v=0
o=- 1746659966980 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 07092115-d151-427e-8722-26c70936b104
m=audio 3480 RTP/SAVP 111 126
c=IN IP4 157.240.19.130
a=rtcp:9 IN IP4 0.0.0.0
a=mid:audio
a=sendrecv
a=msid:07092115-d151-427e-8722-26c70936b104 WhatsAppTrack1
a=rtcp-mux
a=crypto:**************************************************************************
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=maxptime:20
a=ptime:20
a=ssrc:1615009994 cname:WhatsAppAudioStream1

SIP OK (with SDES)

SIP/2.0 200 OK
            CSeq: 31159 INVITE
            Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCA4QkY1MTJCQkNFNTgxMEVFRERFRTUzNTFERkE1MDU0MhwYCzEyMTQ1NTUxODY5FQIAAA
            From: "12195550714" <sip:[email protected]>;tag=8a0f7e65-6e9e-4801-bf92-85c3ef2485d9
            To: <sip:[email protected]>;tag=66596922_c3356d0b_fee164be-566a-4679-a80d-5bfdf1d0aa9e
            Via: SIP/2.0/TLS 157.240.229.209:5061;rport=51830;received=69.171.251.115;branch=z9hG4bKPj3a9f2ad89e4a3df61408aa84f7d9a63e;alias
            Via: SIP/2.0/TLS [2803:6080:f948:9597:d33c:e00:400:0]:5061;branch=z9hG4bKPj3a9f2ad89e4a3df61408aa84f7d9a63e
            Via: SIP/2.0/TLS [2803:6080:f948:9597:1ac5:cdf8:300:0]:63057;rport=63057;received=2803:6080:f948:9597:1ac5:cdf8:300:0;branch=z9hG4bKPj-phic0sbns27DiP0OlrxRxgLtNg4mio7;alias
            Record-Route: <sip:54.172.60.1:5060;lr;r2=on;twnat=sip:69.171.251.115:51830>
            Record-Route: <sip:54.172.60.1:5061;transport=tls;lr;r2=on;twnat=sip:69.171.251.115:51830>
            Record-Route: <sip:wa.meta.vc;transport=tls;lr>
            Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
            Server: Twilio
            Contact: <sip:+172.25.16.223:5060>
            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
            Content-Type: application/sdp
            X-Twilio-CallSid: CAb0d74508fe5fcdf6ec70ea3cf4e9b90b
            Content-Length: 446
            Content-Type: application/sdp
            Content-Length:   446

v=0
o=root 1353670385 1353670385 IN IP4 172.18.164.24
s=Twilio Media Gateway
c=IN IP4 168.86.138.176
t=0 0
m=audio 15822 RTP/SAVP 111 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=crypto:**************************************************************************
a=ptime:20
a=maxptime:20
a=sendrecv

User-initiated calls with Digest Auth (with SDES media)

Meta SIP server supports digest auth for user initiated calls. Your SIP server should respond with digest auth challenge and Meta will resend the SIP INVITE with challenge response. The username used for digest auth is the (normalized) business phone number and the password is generated by Meta and retrievable using the get Call settings endpoint.

First INVITE request from Meta

INVITE sip:[email protected];transport=tls SIP/2.0
Via: SIP/2.0/TLS [2803:6080:f948:9597::]:47237;rport;branch=z9hG4bKPj1e6c665db16b3ecacf32cadb4497fe77;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Via: SIP/2.0/TLS [2803:6080:f948:9597:7253:922a:400:0]:5061;branch=z9hG4bKPj1e6c665db16b3ecacf32cadb4497fe77
Via: SIP/2.0/TLS [2803:6080:f8bc:9272:e488:9927:400:0]:58279;rport=58279;received=2803:6080:f8bc:9272:e488:9927:400:0;branch=z9hG4bKPjr33j97A1mx5J8HWHEy2zIgqZYCCIb4Fb;alias
Max-Forwards: 69
From: "12195550714" <sip:[email protected]>;tag=ece2da15-39e7-4983-ac65-e312f325d94a
To: <sip:[email protected]>
Contact: <sip:[email protected];transport=tls;ob>;isfocus
Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCA2MUI2QUY0MDRCMTUyOTM4QkE5ODEwN0ZGQTAwODkxORwYCzEyMTQ1NTUxODY5FQIAFRoA
CSeq: 9989 INVITE
X-FB-External-Domain: wa.meta.vc
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
User-Agent: Facebook SipGateway
Content-Type: application/sdp
Content-Length:   643

v=0
o=- 1750716867913 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 4e37b099-8aef-45d0-be4f-1cde2ca5a37d
m=audio 3480 RTP/SAVP 111 126
c=IN IP4 57.144.219.49
a=rtcp:9 IN IP4 0.0.0.0
a=mid:audio
a=sendrecv
a=msid:4e37b099-8aef-45d0-be4f-1cde2ca5a37d WhatsAppTrack1
a=rtcp-mux
a=crypto:**************************************************************************
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=maxptime:20
a=ptime:20
a=ssrc:215879358 cname:WhatsAppAudioStream1

407 Response from partner SIP server

SIP/2.0 407 Proxy Authentication required
CSeq: 9989 INVITE
Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCA2MUI2QUY0MDRCMTUyOTM4QkE5ODEwN0ZGQTAwODkxORwYCzEyMTQ1NTUxODY5FQIAFRoA
From: "12195550714" <sip:[email protected]>;tag=ece2da15-39e7-4983-ac65-e312f325d94a
To: <sip:[email protected]>;tag=45065608_c3356d0b_16001fd8-76d2-45f0-bb35-e0441d6dc4a2
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=62080;received=69.171.251.112;branch=z9hG4bKPj1e6c665db16b3ecacf32cadb4497fe77;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:7253:922a:400:0]:5061;branch=z9hG4bKPj1e6c665db16b3ecacf32cadb4497fe77
Via: SIP/2.0/TLS [2803:6080:f8bc:9272:e488:9927:400:0]:58279;rport=58279;received=2803:6080:f8bc:9272:e488:9927:400:0;branch=z9hG4bKPjr33j97A1mx5J8HWHEy2zIgqZYCCIb4Fb;alias
Contact: <sip:+172.25.58.54:5060>
Proxy-Authenticate: Digest realm="sip.twilio.com",nonce="eyOam_8-l5FVugxsyxFRjnlxq9vy1TjQIMB3mBfJuAvB5gV4",opaque="4a6a068be2ca2032a57912b9a2a6adf7",qop="auth"
Content-Length: 0
Content-Length:  0

Second INVITE with authorization from Meta

INVITE sip:[email protected];transport=tls SIP/2.0
Via: SIP/2.0/TLS 31.13.66.215:5061;rport;branch=z9hG4bKPj16be0694dc6763eb66de5ec5f262db03;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Via: SIP/2.0/TLS [2803:6080:f948:9597:7253:922a:400:0]:5061;branch=z9hG4bKPj16be0694dc6763eb66de5ec5f262db03
Via: SIP/2.0/TLS [2803:6080:f8bc:9272:e488:9927:400:0]:58279;rport=58279;received=2803:6080:f8bc:9272:e488:9927:400:0;branch=z9hG4bKPjYp9LqI0D8zJ.wly5wyMyVaH9fUwIU921;alias
Max-Forwards: 69
From: "12195550714" <sip:[email protected]>;tag=ece2da15-39e7-4983-ac65-e312f325d94a
To: <sip:[email protected]>
Contact: <sip:[email protected];transport=tls;ob>;isfocus
Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCA2MUI2QUY0MDRCMTUyOTM4QkE5ODEwN0ZGQTAwODkxORwYCzEyMTQ1NTUxODY5FQIAFRoA
CSeq: 9990 INVITE
X-FB-External-Domain: wa.meta.vc
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
User-Agent: Facebook SipGateway
Proxy-Authorization: Digest username="12145551869", realm="sip.twilio.com", nonce="eyOam_8-l5FVugxsyxFRjnlxq9vy1TjQIMB3mBfJuAvB5gV4", uri="sip:[email protected]", response="b28ed6b8bf1418e3c6eca05ef8c7a0b1", cnonce="TY2SszvYCKitUCBlVLpGiPKMQfmBbj", opaque="4a6a068be2ca2032a57912b9a2a6adf7", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   643

v=0
o=- 1750716867913 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 4e37b099-8aef-45d0-be4f-1cde2ca5a37d
m=audio 3480 RTP/SAVP 111 126
c=IN IP4 57.144.219.49
a=rtcp:9 IN IP4 0.0.0.0
a=mid:audio
a=sendrecv
a=msid:4e37b099-8aef-45d0-be4f-1cde2ca5a37d WhatsAppTrack1
a=rtcp-mux
a=crypto:**************************************************************************
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=maxptime:20
a=ptime:20
a=ssrc:215879358 cname:WhatsAppAudioStream1

SIP OK from partner SIP server

SIP/2.0 200 OK
CSeq: 9990 INVITE
Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCA2MUI2QUY0MDRCMTUyOTM4QkE5ODEwN0ZGQTAwODkxORwYCzEyMTQ1NTUxODY5FQIAFRoA
From: "12195550714" <sip:[email protected]>;tag=ece2da15-39e7-4983-ac65-e312f325d94a
To: <sip:[email protected]>;tag=29360930_c3356d0b_4933dc58-f035-4597-b075-04b19e552329
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=62080;received=69.171.251.112;branch=z9hG4bKPj16be0694dc6763eb66de5ec5f262db03;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:7253:922a:400:0]:5061;branch=z9hG4bKPj16be0694dc6763eb66de5ec5f262db03
Via: SIP/2.0/TLS [2803:6080:f8bc:9272:e488:9927:400:0]:58279;rport=58279;received=2803:6080:f8bc:9272:e488:9927:400:0;branch=z9hG4bKPjYp9LqI0D8zJ.wly5wyMyVaH9fUwIU921;alias
Record-Route: <sip:54.172.60.0:5060;lr;r2=on;twnat=sip:69.171.251.112:62080>
Record-Route: <sip:54.172.60.0:5061;transport=tls;lr;r2=on;twnat=sip:69.171.251.112:62080>
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Contact: <sip:+172.25.43.84:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
X-Twilio-CallSid: CAd4d6e59a356c4d1b0ee85323b2d9dab5
Content-Length: 444
Content-Type: application/sdp
Content-Length:   444

v=0
o=root 477560318 477560318 IN IP4 172.18.156.61
s=Twilio Media Gateway
c=IN IP4 168.86.137.174
t=0 0
m=audio 12710 RTP/SAVP 111 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=crypto:**************************************************************************
a=ptime:20
a=maxptime:20
a=sendrecv

Configure SDES for SRTP key exchange protocol

The Secure Real-time Transport Protocol (SRTP) key exchange is a cryptographic protocol used to securely exchange encryption keys between two parties over an insecure communication channel. You can configure SRTP key exchange to one of two options: DTLS (default) — Industry-standard encrypted key exchange. Recommended. SDES — Plain text key is defined in the SDP which is sent over secure signaling protocol (SIP) or Graph API. When SDES is used, there is no need for STUN, ICE and DTLS which could help shorten the call setup time.

Configure/update SRTP key exchange protocol

Request syntax

POST /&lt;PHONE_NUMBER_ID&gt;/settings

Endpoint parameters

Placeholder	Description	Sample Value
`<PHONE_NUMBER_ID>` Integer	Required The business phone number for which you are updating Calling API settings. Learn more about formatting phone numbers in Cloud API	`+12784358810`

Request body

\{
  "calling": {
    "status": "ENABLED",
    "call_icon_visibility": "DEFAULT"
  . . .
    "srtp_key_exchange_protocol": "DTLS (default) | SDES",
  . . .
  }
. . .
}

Body parameters

Parameter	Description	Sample Value
`srtp_key_exchange_protocol` String	Optional Enable or disable use of SRTP key exchange protocol. Possible values are `SDES` and `DTLS`. Default is `DTLS`. Note: Meta still expects the business side to send the maiden SRTP packet for both user and business initiated calls	`“SDES”`

Success response

\{
  "success": true
}

Error response

View Calling API Error Codes and Troubleshooting for more information View general Cloud API Error Codes here

Get SRTP key exchange protocol

Request syntax

POST /&lt;PHONE_NUMBER_ID&gt;/settings

Endpoint parameters

Placeholder	Description	Sample Value
`<PHONE_NUMBER_ID>` Integer	Required The business phone number for which you are updating Calling API settings. Learn more about formatting phone numbers in Cloud API	`+12784358810`

Response body

\{
  "calling": {
    "status": "ENABLED",
    "call_icon_visibility": "DEFAULT"
  . . .
    "srtp_key_exchange_protocol": "DTLS | SDES",
  . . .
  }
. . .
}

Response parameters

Parameter	Description	Sample Value
`srtp_key_exchange_protocol` String	The type of SRTP key exchange protocol configured for the business phone number queried Possible values are `SDES` and `DTLS`. Default is `DTLS`. Note: If this field has not been explicitly set, it will not be returned.	`“SDES”`

Error response

View Calling API Error Codes and Troubleshooting for more information View general Cloud API Error Codes here

IP Addresses

The IP addresses used for SIP configuration are the same as those listed for the Webhooks in the Cloud API Webhooks IP Addresses section. This reference is solely to indicate the IP addresses to allow-list for SIP traffic. When SIP is enabled, calling related webhooks are not sent.

Troubleshooting

Refer to SIP FAQ for additional SIP specific questions and answers and SIP Errors for SIP specific errors and solutions Did you find this page helpful? Thumbs up icon

ON THIS PAGE Overview Before you get started Signaling and media possible configurations Calling flows using SIP Security How to test if you have a valid TLS certificate Example of valid server cert Example of hostname:port not listening on TLS Example of invalid cert Business-initiated Calls Prerequisites Calling flow User-initiated Calls Prerequisites Calling flow Custom SIP Headers Configure/Update SIP settings on business phone number Request syntax Endpoint parameters Request body Body parameters Success response Error response Get phone number calling settings (SIP) Request syntax Endpoint parameters App Permission Required Response body Include SIP User Password Error response Reset SIP password Sample SIP requests Business-initiated calls (with WebRTC media) Initial SIP INVITE request 407 response from Meta Second SIP INVITE send with authorization Example error response SIP BYE SIP Invite (with SDES) SIP OK (with SDES) User-initiated calls (with WebRTC media) SIP INVITE SIP BYE SIP Invite (with SDES) SIP OK (with SDES) User-initiated calls with Digest Auth (with SDES media) First INVITE request from Meta 407 Response from partner SIP server Second INVITE with authorization from Meta SIP OK from partner SIP server Configure SDES for SRTP key exchange protocol Configure/update SRTP key exchange protocol Request syntax Endpoint parameters Request body Body parameters Success response Error response Get SRTP key exchange protocol Request syntax Endpoint parameters Response body Response parameters Error response IP Addresses Troubleshooting

​Session Initiation Protocol (SIP)

​Overview

​Before you get started

​Signaling and media possible configurations

​Calling flows using SIP

​Security

​How to test if you have a valid TLS certificate

​Example of valid server cert

​Example of hostname:port not listening on TLS

​Example of invalid cert

​Business-initiated Calls

​Prerequisites

​Calling flow

​User-initiated Calls

​Prerequisites

​Calling flow

​Custom SIP Headers

​Configure/Update SIP settings on business phone number

​Request syntax

​Endpoint parameters

​Request body

​Body parameters

​Success response

​Error response

​Get phone number calling settings (SIP)

​Request syntax

​Endpoint parameters

​App Permission Required

​Response body

​Include SIP User Password

​Error response

​Reset SIP password

​Sample SIP requests

​Business-initiated calls (with WebRTC media)

​Initial SIP INVITE request

​407 response from Meta

​Second SIP INVITE send with authorization

​Example error response

​SIP BYE

​SIP Invite (with SDES)

​SIP OK (with SDES)

​User-initiated calls (with WebRTC media)

​SIP INVITE

​SIP BYE

​SIP Invite (with SDES)

​SIP OK (with SDES)

​User-initiated calls with Digest Auth (with SDES media)

​First INVITE request from Meta

​407 Response from partner SIP server

​Second INVITE with authorization from Meta

​SIP OK from partner SIP server

​Configure SDES for SRTP key exchange protocol

​Configure/update SRTP key exchange protocol

​Request syntax

​Endpoint parameters

​Request body

​Body parameters

​Success response

​Error response

​Get SRTP key exchange protocol

​Request syntax

​Endpoint parameters

​Response body

​Response parameters

​Error response

​IP Addresses

​Troubleshooting

Session Initiation Protocol (SIP)

Overview

Before you get started

Signaling and media possible configurations

Calling flows using SIP

Security

How to test if you have a valid TLS certificate

Example of valid server cert

Example of hostname:port not listening on TLS

Example of invalid cert

Business-initiated Calls

Prerequisites

Calling flow

User-initiated Calls

Prerequisites

Calling flow

Custom SIP Headers

Configure/Update SIP settings on business phone number

Request syntax

Endpoint parameters

Request body

Body parameters

Success response

Error response

Get phone number calling settings (SIP)

Request syntax

Endpoint parameters

App Permission Required

Response body

Include SIP User Password

Error response

Reset SIP password

Sample SIP requests

Business-initiated calls (with WebRTC media)

Initial SIP INVITE request

407 response from Meta

Second SIP INVITE send with authorization

Example error response

SIP BYE

SIP Invite (with SDES)

SIP OK (with SDES)

User-initiated calls (with WebRTC media)

SIP INVITE

SIP BYE

SIP Invite (with SDES)

SIP OK (with SDES)

User-initiated calls with Digest Auth (with SDES media)

First INVITE request from Meta

407 Response from partner SIP server

Second INVITE with authorization from Meta

SIP OK from partner SIP server

Configure SDES for SRTP key exchange protocol

Configure/update SRTP key exchange protocol

Request syntax

Endpoint parameters

Request body

Body parameters

Success response

Error response

Get SRTP key exchange protocol

Request syntax

Endpoint parameters

Response body

Response parameters

Error response

IP Addresses

Troubleshooting