> ## Documentation Index > Fetch the complete documentation index at: https://whatsapp-docs.kap.so/llms.txt > Use this file to discover all available pages before exploring further. # Sip # 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](https://l.facebook.com/l.php?u=https%3A%2F%2Fdatatracker.ietf.org%2Fdoc%2Fhtml%2Frfc3261\&h=AT04zUr58l0YkXmB7_k3AbktrDm8VIc_xfvtonr9Qlfa9p76J9Q2sDTCCNnTuFUKUo9vErBEP26tb6_TpDG89vujKkq5zO-RVc5cA_nDW_VlLF6-wMWtk9NxD8E4bObWaLcUq62fz7C49Q)) 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling#step-1--prerequisites) 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/support/health-status) with `PHONE_NUMBER_ID` Your app mode is “Live”, not “Development”. You have a standards compliant third party SIP server that supports [TLS](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#security) 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#configure-update-sip-settings-on-business-phone-number)) | SIP (needs explicit [enablement](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#configure-update-sip-settings-on-business-phone-number)) | | Signaling transport | HTTPS | TLS | TLS | | Media protocol | WebRTC (ICE + DTLS\* + SRTP) | WebRTC (ICE + DTLS + SRTP) | [SDES](https://l.facebook.com/l.php?u=https%3A%2F%2Fdatatracker.ietf.org%2Fdoc%2Fhtml%2Frfc4568\&h=AT04zUr58l0YkXmB7_k3AbktrDm8VIc_xfvtonr9Qlfa9p76J9Q2sDTCCNnTuFUKUo9vErBEP26tb6_TpDG89vujKkq5zO-RVc5cA_nDW_VlLF6-wMWtk9NxD8E4bObWaLcUq62fz7C49Q) SRTP (needs explicit [enablement](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#configure-sdes-for-srtp-key-exchange-protocol)) | | 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#configure-update-sip-settings-on-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.](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#get-phone-number-calling-settings--sip-) ### 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#configure-update-sip-settings-on-business-phone-number) 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/user-call-permissions) [Retrieve Meta generated SIP password](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#include-sip-user-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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#business-initiated-calls--with-webrtc-media-) 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:[+11234567890@wa.meta.vc](mailto:+11234567890@wa.meta.vc);transport=tls’ This request will fail with an “SIP 407 Proxy Authentication required” message. Send a 2nd [SIP INVITE](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#business-initiated-calls--with-webrtc-media-) with proper Authorization header as per [RFC 3261](https://l.facebook.com/l.php?u=https%3A%2F%2Fdatatracker.ietf.org%2Fdoc%2Fhtml%2Frfc3261%23section-22\&h=AT04zUr58l0YkXmB7_k3AbktrDm8VIc_xfvtonr9Qlfa9p76J9Q2sDTCCNnTuFUKUo9vErBEP26tb6_TpDG89vujKkq5zO-RVc5cA_nDW_VlLF6-wMWtk9NxD8E4bObWaLcUq62fz7C49Q). 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#get-phone-number-calling-settings--sip-) 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#business-initiated-calls--with-webrtc-media-) 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#include-sip-user-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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#user-initiated-calls--with-webrtc-media-) to the SIP server [configured on the business phone number](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#configure-update-sip-settings-on-business-phone-number) You respond with [SIP digest auth challenge](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#user-initiated-calls-with-digest-auth--with-sdes-media-) (recommended) or [SIP OK](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#user-initiated-calls--with-webrtc-media-) and pass in an SDP answer If you are not receiving SIP INVITE from Meta, refer to [SIP specific FAQ](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/faq#session-initiation-protocol--sip--faq) to troubleshoot further [View sample SIP requests](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#sample-sip-requests) [Learn more about Session Description Protocol (SDP)](https://l.facebook.com/l.php?u=https%3A%2F%2Fwww.rfc-editor.org%2Frfc%2Frfc8866.html\&h=AT04zUr58l0YkXmB7_k3AbktrDm8VIc_xfvtonr9Qlfa9p76J9Q2sDTCCNnTuFUKUo9vErBEP26tb6_TpDG89vujKkq5zO-RVc5cA_nDW_VlLF6-wMWtk9NxD8E4bObWaLcUq62fz7C49Q) [View example SDP structures](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/reference#sdp-overview-and-sample-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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/call-button-messages-deep-links#send-interactive-message-with-a-whatsapp-call-button) | | x-wa-meta-deeplink-payload | Optional; String | Present when user-initiates a call from call deeplink that has business specified payload. [Learn more](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/call-button-messages-deep-links#send-payload-data-in-call-deeplink) | ## 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 /<PHONE_NUMBER_ID>/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](https://developers.facebook.com/documentation/business-messaging/whatsapp/business-phone-numbers/phone-numbers) | `+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](https://l.facebook.com/l.php?u=https%3A%2F%2Fdatatracker.ietf.org%2Fdoc%2Fhtml%2Frfc4904\&h=AT04zUr58l0YkXmB7_k3AbktrDm8VIc_xfvtonr9Qlfa9p76J9Q2sDTCCNnTuFUKUo9vErBEP26tb6_TpDG89vujKkq5zO-RVc5cA_nDW_VlLF6-wMWtk9NxD8E4bObWaLcUq62fz7C49Q))
sip:[+1234567890@sip.example.com](mailto:+1234567890@sip.example.com);
tgrp=wacall;
trunk-context=byoc.example.com
This example has two user parameters for tgrp, and trunk-context. | `
"servers": [
   {
      "hostname": SIP_SERVER_HOSTNAME
      "port": SIP_SERVER_PORT,
      "request_uri_user_params": {
         "KEY1": "VALUE1",
         "KEY2": "VALUE2",
      }
   }
]
` | ### Success response ``` \{   "success": true } ``` ### Error response [View Calling API Error Codes and Troubleshooting for more information](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/troubleshooting) [View general Cloud API Error Codes here](https://developers.facebook.com/documentation/business-messaging/whatsapp/support/error-codes) ## 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 /<PHONE_NUMBER_ID>/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](https://developers.facebook.com/documentation/business-messaging/whatsapp/business-phone-numbers/phone-numbers) | `+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": <APP_ID_THAT_CONFIGURED_THIS_SIP_SERVER>,\           "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 /<PHONE_NUMBER_ID>/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": <APP_ID_THAT_CONFIGURED_THIS_SIP_SERVER>,\           "hostname": "sip.example.com",\           "sip_user_password": "{SIP_USER_PASSWORD}"\         }\       ]     }   } } ``` ### Error response [View Calling API Error Codes and Troubleshooting for more information](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/troubleshooting) [View general Cloud API Error Codes here](https://developers.facebook.com/documentation/business-messaging/whatsapp/support/error-codes) ## 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#include-sip-user-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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#include-sip-user-password) to note the new password ## Sample SIP requests ### Business-initiated calls (with WebRTC media) #### Initial SIP INVITE request ``` INVITE sip:+12195550714@wa.meta.vc;transport=tls SIP/2.0 Record-Route: 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: ;tag=Kc9QZg4496maQ To: Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba CSeq: 96743476 INVITE Contact: 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: ;party=calling;screen=yes;privacy=off Content-Type: application/sdp Content-Length: 2427 <> ``` #### 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: Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba From: ;tag=Kc9QZg4496maQ To: ;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:+12195550714@wa.meta.vc;transport=tls SIP/2.0 Record-Route: 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: ;tag=Kc9QZg4496maQ To: Call-ID: dc2c5b33-1b81-43ee-9213-afb56f4e56ba CSeq: 96743477 INVITE Contact: 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:+12195550714@wa.meta.vc;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: ;party=calling;screen=yes;privacy=off Content-Type: application/sdp Content-Length: 2427 <> ``` #### 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: Record-Route: Call-ID: 91578781-44f1-4268-9a7f-d7efec1abf72 From: ;tag=3a63b370-a697-4a5a-82b4-e8105e23f176 To: ;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:+5559800000693@wa.meta.vc;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: Route: Max-Forwards: 69 From: ;tag=NcKQ6mtDKSDQB To: "5559800000693" ;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:+12195550714@wa.meta.vc;transport=tls SIP/2.0 Record-Route: Record-Route: CSeq: 2 INVITE From: "12145551869" ;tag=28460006_c3356d0b_5cdada8c-cbf0-4369-b02d-cc97d3c36f2b To: Max-Forwards: 66 P-Asserted-Identity: Min-SE: 120 Call-ID: f304a1d2cafb8139c1f9ff93a7733586@0.0.0.0 Contact: "12145551869" 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:+12195550714@wa.meta.vc;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: Record-Route: Record-Route: Record-Route: Call-ID: f304a1d2cafb8139c1f9ff93a7733586@0.0.0.0 From: "12145551869" ;tag=28460006_c3356d0b_5cdada8c-cbf0-4369-b02d-cc97d3c36f2b To: ;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: ;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:+17015558857@meta-voip.example.com;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" ;tag=bbf1ad6e-79bb-4d9c-8a2c-094168a10bea To: Contact: ;isfocus Call-ID: outgoing:wacid.HBgLMTIxOTU1NTA3MTQVAgASGCAzODg1NTE5NEU1NTBEMTc1RTFFQUY5NjNCQ0FCRkEzRhwYCzE3MDE1NTU4ODU3FQIAAA== CSeq: 2824 INVITE Route: 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:+5559800000693@wa.meta.vc;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: Route: Max-Forwards: 69 From: ;tag=NcKQ6mtDKSDQB To: "5559800000693" ;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:+12145551869@meta-voip.example.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS [2803:6080:f948:9597::]:57363;rport;branch=z9hG4bKPj3a9f2ad89e4a3df61408aa84f7d9a63e;alias Record-Route: Record-Route: 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" ;tag=8a0f7e65-6e9e-4801-bf92-85c3ef2485d9 To: Contact: ;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" ;tag=8a0f7e65-6e9e-4801-bf92-85c3ef2485d9 To: ;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: Record-Route: Record-Route: Record-Route: Server: Twilio Contact: 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/sip#include-sip-user-password). #### First INVITE request from Meta ``` INVITE sip:+12145551869@meta-voip.example.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS [2803:6080:f948:9597::]:47237;rport;branch=z9hG4bKPj1e6c665db16b3ecacf32cadb4497fe77;alias Record-Route: Record-Route: 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" ;tag=ece2da15-39e7-4983-ac65-e312f325d94a To: Contact: ;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" ;tag=ece2da15-39e7-4983-ac65-e312f325d94a To: ;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: 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:+12145551869@meta-voip.example.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS 31.13.66.215:5061;rport;branch=z9hG4bKPj16be0694dc6763eb66de5ec5f262db03;alias Record-Route: Record-Route: 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" ;tag=ece2da15-39e7-4983-ac65-e312f325d94a To: Contact: ;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:+12145551869@meta-voip.example.com", 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" ;tag=ece2da15-39e7-4983-ac65-e312f325d94a To: ;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: Record-Route: Record-Route: Record-Route: Contact: 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 /<PHONE_NUMBER_ID>/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](https://developers.facebook.com/documentation/business-messaging/whatsapp/business-phone-numbers/phone-numbers) | `+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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/troubleshooting) [View general Cloud API Error Codes here](https://developers.facebook.com/documentation/business-messaging/whatsapp/support/error-codes) ### Get SRTP key exchange protocol #### Request syntax ``` POST /<PHONE_NUMBER_ID>/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](https://developers.facebook.com/documentation/business-messaging/whatsapp/business-phone-numbers/phone-numbers) | `+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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/troubleshooting) [View general Cloud API Error Codes here](https://developers.facebook.com/documentation/business-messaging/whatsapp/support/error-codes) ## 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/webhooks/overview#ip-addresses). 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](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/faq#session-initiation-protocol--sip--faq) for additional SIP specific questions and answers and [SIP Errors](https://developers.facebook.com/documentation/business-messaging/whatsapp/calling/troubleshooting#sip-errors) for SIP specific errors and solutions Did you find this page helpful? ![Thumbs up icon](https://static.xx.fbcdn.net/rsrc.php/yR/r/OEXJ0_DJeZv.svg) ![Thumbs down icon](https://static.xx.fbcdn.net/rsrc.php/yb/r/qKPgNVNeatU.svg) 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 ***