Integration Examples | Developer Documentation Integration ExamplesUpdated: Feb 25, 2026This guide explains integration of common VoIP platforms with WhatsApp Business Calling API.This guide is for information purposes only with no support or warranties of any kind from Meta or any vendor. There are many ways to integrate and the guide explains just one way exclusively for illustrative purposes. Asterisk using SIP OverviewThis guide explains how to set up WhatsApp Business Calling API using SIP signaling with Asterisk⁠, an open-source PBX (Private Branch Exchange). You’ll learn how to configure your Asterisk server, connect SIP phones, and handle both incoming and outgoing WhatsApp calls. User-initiated calls The WhatsApp user dials the business number.The call is received by Asterisk and routed through an IVR, prompting the user to enter an extension, registered to the same Asterisk server.The call is then connected to the specified extension. Business-initiated calls The business agent/user registers with Asterisk using SIP credentials (see “Configuring a VoIP Phone” section).The business user dials the b2c-sip (business to consumer) extension, which is handled by an IVR. The IVR prompts for the WhatsApp number to call.The call is then connected to the WhatsApp user.The WA to Asterisk leg uses SDES for media encryption key exchange and opus for audio codecThe Asterisk to SIP UA leg uses SDES for media encryption key exchange and opus or G711 for audio codec Prerequisites Asterisk Deployment: Asterisk is deployed (for example, on a public cloud instance)Operating System: Any OS compatible for Asterisk. For example, CentOS 9Domain: Asterisk server is reachable via a public domain with valid certificateWhatsApp Business API: A WhatsApp business phone number is registered and calling is enabled.SIP Support: SIP is enabled on the WhatsApp Business NumberSDES: SDES is enabled on the WhatsApp Business Number Building and installing AsteriskRefer to https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Building-and-Installing-Asterisk/⁠This guide was tested using Asterisk version 22.5.2 Asterisk configurationThese configuration files are placed under /etc/asterisk/ extensions.confReplace the following placeholders with actual values {wa-business-phone-number}: WhatsApp Business Phone Number{asterisk-sip-server-dns}: DNS name of your Asterisk SIP serverincoming_welcome: incoming_welcome.wav (not provided) place this file under /var/lib/asterisk/soundsoutgoing_welcome: outgoing_welcome.wav (not provided) place this file under /var/lib/asterisk/sounds [c2b-sub-dial] exten => s,1,NoOp() same => n,Read(Digits,incoming_welcome,0,,5,500) same => n,Dial(PJSIP/${Digits}) same => n,Hangup()[whatsapp] exten => _10XX,1,NoOp() same => n,Dial(PJSIP/${EXTEN}) same => n,Hangup();Extensionfor B2C business call through Meta SIP gateway exten => b2c-sip,1,NoOp() same => n,Read(Digits,outgoing_welcome,0,,5,500) same => n,Dial(PJSIP/whatsapp/sip:${Digits}@wa.meta.vc);Extension to handle incoming invite requests fromMeta SIP gateway to @ exten => _+,1,Goto(c2b-sub-dial,s,1) Pjsip.confReplace the following placeholders with actual values {wa-business-phone-number} : the business phone number{local-net}: local network of the Asterisk server{external-media-address}: Public IP of the Asterisk server media{external-signaling-address}: Public IP of the Asterisk server signaling{sip-ua-password}: Chosen SIP User Agent password{domain-name}: domain name assigned to the Asterisk serverCertificate files should be placed under /var/lib/asterisk/certs/fullchain.cer /var/lib/asterisk/certs/cer.key [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/var/lib/asterisk/certs/fullchain.cer priv_key_file=/var/lib/asterisk/certs/cer.key method=sslv23 allow_wildcard_certs=yes external_media_address={external-media-address};External address for SIP signalling external_signaling_address={external-signaling-address};Network to consider local used for NAT purposes local_net={local-net}[sdes_endpointtemplate](!) type=endpoint context=whatsapp disallow=all allow=OPUS direct_media=no rtp_symmetric=yes force_rport=yes rewrite_contact=no media_use_received_transport=yes media_encryption=sdes [authtemplate](!) type=auth auth_type=userpass password={sip-ua-password}[aortemplate](!) type=aor max_contacts=1 remove_existing=yes [aoridentitytemplate](!) type=identify match_header=X-FB-External-Domain: wa.meta.vc ;SDES users [1000](sdes_endpointtemplate) auth=1000_auth aors=1000[1000_auth](authtemplate) username=1000[1000](aortemplate)[1000](aoridentitytemplate) endpoint=1000[1001](sdes_endpointtemplate) auth=1001_auth aors=1001[1001_auth](authtemplate) username=1001[1001](aortemplate)[1001](aoridentitytemplate) endpoint=1001[1002](sdes_endpointtemplate) auth=1002_auth aors=1002[1002_auth](authtemplate) username=1002[1002](aortemplate)[1002](aoridentitytemplate) endpoint=1002[1003](sdes_endpointtemplate) auth=1003_auth aors=1003[1003_auth](authtemplate) username=1003[1003](aortemplate)[1003](aoridentitytemplate) endpoint=1003[1004](sdes_endpointtemplate) auth=1004_auth aors=1004[1004_auth](authtemplate) username=1004[1004](aortemplate)[1004](aoridentitytemplate) endpoint=1004[1005](sdes_endpointtemplate) auth=1005_auth aors=1005[1005_auth](authtemplate) username=1005[1005](aortemplate)[1005](aoridentitytemplate) endpoint=1005;This endpoint maps to an IVR for C2B calls [c2b-sip](sdes_endpointtemplate)[c2b-sip](aortemplate)[c2b-sip] type=identify endpoint=c2b-sip match_header=X-FB-External-Domain: wa.meta.vc ;special endpoint forMeta SIP Gateway integration ;This endpoint maps to an IVR for B2C calls [b2c-sip](sdes_endpointtemplate)[b2c-sip](aortemplate)[whatsapp](sdes_endpointtemplate) type=endpoint transport=transport-tls disallow=all allow=opus,ulaw,alaw aors=whatsapp from_user={wa-business-phone-number} from_domain={domain-name} outbound_auth=whatsapp [whatsapp] type=aor contact=sip:wa.meta.vc [whatsapp] type=identify endpoint=whatsapp [whatsapp] type=auth auth_type=digest password={meta-sip-user-password} username={wa-business-phone-number} realm=* rtp.conf [general];Hostnameor address for the STUN server used for determining the external ; IP address and port an RTP session can be reached at.The port number is; optional.If omitted default value of 3478 will be used.This option is; disabled bydefault.Name resolution occurs at load time,andif DNS is; used, name resolution will occur repeatedly after the TTL expires.;;for example stundaddr=mystun.server.com:3478; stunaddr=stun.l.google.com:19302 rtpstart=10000 rtpend=60000 Configuring a VoIP phoneDownload and install a softphone client (for example, Linphone⁠) for testing both business-initiated and user-initiated calls. Account setup Select an extension to register as a SIP UA (extensions 1001–1005).Open Preferences.Under “SIP Accounts,” click “Add account.”Enter the following details: SIP Address: for example, sip:1001@{asterisk-sip-server-dns}SIP Server Address: for example, sip:{asterisk-sip-server-dns};transport=tlsTransport: TLSDisable ICEEnable AVPFDisable “Publish presence information”Confirm and save the account.Enter the password when prompted (viz. {sip-ua-password})Once connected, return to Preferences and select the “Audio” tab. Enable all audio codecs.In the “Calls and Chat” tab: Select “Encryption”Choose “SRTP-SDES”Enable “Encryption is mandatory”Confirm settings Final checklist Double-check all configuration files for correct numbers, passwords, and domain names.Make sure your firewall allows SIP (5061/TLS) and RTP (10000-20000) ports.For more details on SIP password setup, see the WhatsApp Cloud API documentation. Troubleshooting Cannot register SIP UAConfirm that the SIP URL is correct and the domain is pointing to the Asterisk server. Run host {domain-name} to verify that the IP address points to the Asterisk server. Not receiving ACK from Meta OR Business audio stops around 30s OR Meta returns 404 response to BYEFor a user initiated call, Meta sends a SIP INVITE to your SIP server which then responds with 200 OK. Meta acks your 200 OK with an ACK but you never receive this ACK. So your SIP server keeps resending the 200 OK and ultimately the SIP dialog is terminated due to ACK timeout (typically 32s).The most likely cause for this problem is incorrect Record-Route headers in your 200 OK to Meta. The 200 OK response is supposed to not modify the Record-Route headers included in the original Meta’s INVITE. Your SIP server can add new Record-Route headers but cannot modify those present in our INVITEThe solution to this problem is to change rewrite_contact=yes to rewrite_contact=no on the WhatsApp endpoint configuration in pjsip.conf file. After this make sure your 200 OK has following headers as the last 2 in the list of Record-Route headersThis problem is hard to detect or diagnose. Even with this bug, the call will get connected and media will flow both sides but around 32s later, your SIP server will terminate the call which won’t be propagated to WhatsApp client because your BYE request has incorrect Route headers. So WA user stops hearing business audio around 32s. Record-Route:Record-Route: FreeSWITCH using SIP OverviewThis guide explains how to set up WhatsApp Business Calling API using SIP signaling with FreeSWITCH⁠, an open-source communication framework. You’ll learn how to configure your FreeSWITCH server, connect SIP phones, and handle both user-initiated and business-initiated WhatsApp calls. User-initiated calls The WhatsApp user dials the business number.The call is received by FreeSWITCH and routed through an IVR, which prompts the user to enter an agent’s extension registered on the same FreeSWITCH server.Once the extension is entered, the call is connected to the specified recipient agent. Business-initiated calls The business agent or user registers with FreeSWITCH using SIP credentials (see the Configuring a VoIP Phone section for details).The business user dials the b2c-sip (business-to-consumer) extension, which is managed by an IVR. The IVR then prompts for the WhatsApp number to call.After the number is entered, the call is connected to the WhatsApp user via SIP.The WA to FreeSWITCH leg uses SDES for media encryption key exchange with Opus as the audio codec. FreeSWITCH - SIP UA leg uses SDES for media encryption key exchange with Opus or G.711 audio codecs Prerequisites FreeSWITCH Deployment: FreeSWITCH is deployed (for example, on a public cloud instance)Operating System: Any OS compatible with FreeSWITCH. For example, CentOS 9Domain: FreeSWITCH server is reachable via a public domain with a valid certificateWhatsApp Business API: A WhatsApp business phone number is registered and calling is enabled.SIP Support: SIP is enabled on the WhatsApp Business Number Note: FreeSWITCH is configured to listen on 5081 for TLSSDES: SDES is enabled on the WhatsApp Business Number Building and installing FreeSWITCHRefer to https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Installation/⁠This guide was tested using FreeSWITCH version 1.10.12. FreeSWITCH uses sofia (an open-source SIP user agent library). Sofia v1.13.17 was used for this guide FreeSWITCH configurationThese configuration files are placed under /usr/share/freeswitch/etc/freeswitchwa-biz-api-dialplan.xmlPlace the dial plan under /usr/share/freeswitch/etc/freeswitch/dialplan/default/wa-biz-api-dialplan.xml Audio files should be placed under /usr/share/freeswitch/sounds (not provided) incoming_welcome.wavIncoming_invalid.wavoutgoing_welcome.wavoutgoing_invalid.wavwhatsapp.xmlThis file configures the WhatsApp gateway, copy the file to /usr/share/freeswitch/etc/freeswitch/sip_profiles/external/whatsapp.xml Replace the following placeholders with actual values {phone-number}: WhatsApp Business Phone Number{meta-sip-password}: SIP password issued by Meta. For more details on SIP password setup, see the WhatsApp Cloud API documentation.{domain-name}: DNS name of your FreeSWITCH SIP serveracl.conf.xmlOpen /usr/share/freeswitch/etc/freeswitch/autoload_configs/acl.conf.xmlAdd the following list under network-lists element vars.xmlModify /usr/share/freeswitch/etc/freeswitch/vars.xml Add line under Replacewith(substitute {sip_ua_password}with your password)Replacewith(substitute {domain-name}with your FreeSWITCH SIP server DNS) with (substitute {external-ip} with your FreeSWITCH public ip) Replace with (substitute {external-ip} with your FreeSWITCH public ip) internal.xmlModify /usr/share/freeswitch/etc/freeswitch/sip_profiles/internal.xml Look for: Replace it with external.xml Modify /usr/share/freeswitch/etc/freeswitch/sip_profiles/external.xml ReplacewithReplacewithReplacewithMake sure certificates are placed under /usr/share/freeswitch/etc/freeswitch/certs Final checklist Double-check all configuration files for correct numbers, passwords, and domain names.Make sure your firewall allows SIP (5081/TLS) and RTP (10000-20000) ports.For more details on SIP password setup, see the WhatsApp Cloud API documentation. Troubleshooting Cannot register SIP UAConfirm that the SIP URL is correct and the domain is pointing to the FreeSWITCH server. Run host {domain-name} to verify that the IP address points to the FreeSWITCH server. Trace SIP messagesStart CLI (/usr/share/freeswitch/bin/fs_cli) to view SIP messages FreeSWITCH using Graph API with Janus OverviewThis guide explains how to set up WhatsApp Business Calling API using WhatsApp Cloud API signaling with FreeSWITCH⁠, an open-source communication framework and Janus⁠, a general purpose WebRTC server. You’ll learn how to configure your FreeSWITCH server, connect SIP phones, and handle both incoming and outgoing WhatsApp calls. User-initiated calls The WhatsApp user dials the business number.The call is received by Webhook server which forwards it to FreeSWITCH server via Janus SIP plugin.The call is received by FreeSWITCH and routed through an IVR, prompting the user to enter an extension, registered to the same FreeSWITCH server.The call is then connected to the specified extension. Business-initiated calls The business agent/user registers with FreeSWITCH using SIP credentials (see “Configuring a VoIP Phone” section).The business user dials the b2c-sip (business to consumer) extension, which is handled by an IVR. The IVR prompts for the WhatsApp number to call.FreeSWITCH bridges the call to extension registered to Janus SIP plugin which translates it to an API request to MetaThe call is then connected to the WhatsApp user.The Janus server sits between WA and FreeSWITCH and converts media from WA (WebRTC complaint with DTLS key exchange) to FreeSWITCH negotiated media (SDES key exchange).FreeSWITCH - Sip UA will be using SDES for media encryption key exchange and opus or G711 for audio codec Prerequisites FreeSWITCH Deployment: FreeSWITCH is deployed (for example, on a public cloud instance)Janus Deployment: Can be deployed on the same machine as FreeSWITCHOperating System: Any OS compatible with FreeSWITCH. For example, CentOS 9Domain: FreeSWITCH server and Webhook server are reachable via a public domain with valid certificateWhatsApp Business API: A WhatsApp business phone number is registered and calling is enabled.Webhooks: Configure Webhook callback URL pointing to domain name of the Webhook server Integration with Cloud API signalingYou will need to implement an integration module which sits between WA and Janus and translates Cloud API Signalling messages to Janus SIP plugin messages and vice versa.You will need A webhook server to receive calls webhook events from MetaA Graph API module to send call messages to MetaAn implementation of Janus SIP plugin⁠ to connect to Janus. The Janus plugin implementation will connect to FreeSWITCH using extension 1000 which is reserved for bridgingBusiness initiated calls The module will receive a SIP INVITE via Janus SIP plugin on extension 1000. The SIP INVITE is converted to a Graph API request. The SDP received in the SIP INVITE is sent verbatim as the SDP offer to WA via the Graph API callWhen the call is accepted by the WA user, an accepted webhook is received. On receiving the webhook, the Janus SIP Plugin accepts the SIP INVITE passing the answer SDP in the connect webhookUser Initiated calls The webhook server receives an incoming call via a webhook message containing the offer SDP. On receiving the call invite, the Janus SIP plugin sends an invite to FreeSWITCH via extension 1000. The destination extension is c2b-sip.When the Janus SIP plugin receives the SIP 200 OK, a Graph API accept call request is sent to Meta to accept the incoming call by passing the SDP received as part of SIP answer Building and installing JanusRefer to https://github.com/meetecho/janus-gateway⁠ This guide was tested using version 1.2.3 Janus configurationjanus.jcfgModify janus.jcfg which can be found at /usr/share/janus/etc/janus/janus.jcfg Set nat_1_1_mapping to the public IP of the Janus ServerTo start Janus /usr/share/janus/bin/janus --debug-level=6--libnice-debug=on -S stun.l.google.com:19302--log-file=/var/log/janus.log --config=/usr/share/janus/etc/janus/janus.jcfg Building and installing FreeSWITCHRefer to https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Installation/⁠This guide was tested using FreeSWITCH version 1.10.12. FreeSWITCH uses sofia (an open-source SIP user agent library). Sofia v1.13.17 was used for this guideFreeSWITCH Configuration These configuration files are placed under /usr/share/freeswitch/etc/freeswitchwa-biz-api-dialplan.xmlPlace the dial plan under /usr/share/freeswitch/etc/freeswitch/dialplan/default/wa-biz-api-dialplan.xml Audio files should be placed under /usr/share/freeswitch/sounds (not provided) incoming_welcome.wavIncoming_invalid.wavoutgoing_welcome.wavoutgoing_invalid.wavinternal.xmlModify /usr/share/freeswitch/etc/freeswitch/sip_profiles/internal.xml Look for: Replace it with Configuring a VoIP phoneRefer to the earlier section Final checklist Double-check all configuration files for correct numbers, passwords, and domain names.Make sure your firewall allows SIP (5061/TLS) and RTP (10000-20000) ports.For more details on SIP password setup, see the WhatsApp Cloud API documentation. Troubleshooting Cannot register SIP UAConfirm that the SIP URL is correct and the domain is pointing to the FreeSWITCH server. Run host {domain-name} to verify that the IP address points to the FreeSWITCH server. Trace SIP messagesStart CLI (/usr/share/freeswitch/bin/fs_cli) to view SIP messages Asterisk using Graph API with RtpEngine OverviewThis guide explains how to set up WhatsApp Business Calling API using WhatsApp Cloud API signaling with Asterisk⁠, an open-source PBX (Private Branch Exchange) and RtpEngine⁠, an open-source proxy used for relaying, manipulating, and controlling RTP streams. You’ll learn how to configure your Asterisk server, connect SIP phones, and handle both incoming and outgoing WhatsApp calls. User-initiated calls The WhatsApp user dials the business number.The call is received by the Webhook server which after bridging media using RtpEngine, forwards it to Asterisk using SIP.The call is received by Asterisk and routed through an IVR, prompting the user to enter an extension, registered to the same Asterisk server.The call is then connected to the specified extension. Business-initiated calls The business agent/user registers with Asterisk using SIP credentials (see “Configuring a VoIP Phone” section).The business user dials the b2c-sip (business to consumer) extension, which is handled by an IVR. The IVR prompts for the WhatsApp number to call.Asterisk bridges the call to extension registered by the integration module (see “Integration with Cloud API Signalling”)On receiving the call, the integration module bridges the media using RtpEngine and then translates it to an API request to MetaThe call is then connected to the WhatsApp user.RtpEngine acts as a media proxy and sits between the media stream of WA (WebRTC complaint with DTLS key exchange) and Asterisk (SDES key exchange) Prerequisites Asterisk Deployment: Asterisk is deployed (for example, on a public cloud instance)RtpEngine Deployment: Can be deployed on the same machine as AsteriskOperating System: Any OS compatible with Asterisk and RtpEngine. For example, CentOS 9Domain: Asterisk server and Webhook server are reachable via a public domain with valid certificateWhatsApp Business API: A WhatsApp business phone number is registered and calling is enabled.Webhooks: Configure Webhook callback URL pointing to domain name of the Webhook server Integration with Cloud API signalingYou will need to implement an integration module that acts as a bridge between WhatsApp and Asterisk. This module will: Translate Cloud API Signaling messages from WhatsApp to SIP for Asterisk, and vice versaUse SIP signaling for communication between the SIP UA inside the module and AsteriskBridge the media between WhatsApp and Asterisk via RtpEngineYou will need following components, which are part of the integration module for the purpose of this setup Webhook Server: Receives call webhook events from Meta (WhatsApp Cloud API)Graph API client: Sends call-related requests to Meta using the Graph APISIP User Agent (UA) such as PJSIP: Connects to Asterisk using extension 1000, which is reserved for bridging calls between WhatsApp and Asterisk.RtpEngineClient: To control RtpEngine via ng control protocol⁠ for bridging media Business initiated calls Business agent registered to the same Asterisk server dials b2c-sip extension to initiate a call to WhatsApp userThe extension prompts the business agent to enter WA user’s phone numberAsterisk sends a SIP INVITE request to extension 1000 with a custom header containing the dialed WA user phone numberThe SIP UA inside the module would’ve registered at extension 1000 and hence receives the SIP INVITE from AsteriskThe SDP included in the SIP INVITE is sent to RtpEngine which returns a new SDPThe new SDP is included in the Graph API request to initiate a new callWhen the WhatsApp user accepts the call, an “accepted” webhook is receivedUpon receiving this webhook, the answer SDP received in the webhook is sent to RtpEngine which returns a new SDPThe SIP UA accepts the original SIP INVITE (step 3), passing along the new SDP received from RtpEngineThe call is now bridged between WA user, RtpEngine, and AsteriskUser Initiated calls The webhook server inside the module receives an incoming call webhook from Meta, which includes the offer SDPUpon receiving this call invite, the SDP included in the offer is sent to RtpEngine which returns a new SDPThe SIP UA inside the module sends a SIP INVITE to Asterisk using extension 1000 passing the new SDP from RtpEngine in the SIP INVITE. The destination extension is c2b-sip.The extension prompts WA user to dial the extension of the business agent to connect toAsterisk dials the specified extension and waits for an answerAfter the agent answers the call, Asterisk sends SIP 200 OK to the SIP UA extension 1000 inside the module. The SDP in SIP 200 OK is sent to RtpEngine which returns a new SDPA Graph API request is sent to Meta to accept the incoming call, with the new SDP received from RtpEngine Building and installing AsteriskRefer to https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Building-and-Installing-Asterisk/This guide was tested using Asterisk version 22.5.2 Building and installing RtpEngineRefer to https://github.com/sipwise/rtpengine⁠ to build and install RtpEngine This guide was tested using RtpEngine version 13.3.1.4Refer to https://rtpengine.readthedocs.io/en/latest/ng_control_protocol.html for details on ng control protocolTo start RtpEngine run /usr/bin/rtpengine --listen-ng={local-ip}:22222--interface={local-ip}\!{public-ip}-f -EReplace {local-ip} with the local IP of the RtpEngine server{public-ip} with the public IP of the RtpEngine serverAsterisk Configuration These configuration files are placed under /etc/asterisk/extensions.confReplace the following placeholders with actual values incoming_welcome: incoming_welcome.wav (not provided) place this file under /var/lib/asterisk/soundsoutgoing_welcome: outgoing_welcome.wav (not provided) place this file under /var/lib/asterisk/sounds [handler];Set headers on callee channel exten => addheader,1,Set(PJSIP_HEADER(add,X-WhatsApp-Number)=${DIGITS}) same => n,Return()[default] exten => _10XX,1,NoOp() same => n,Dial(PJSIP/${EXTEN}) same => n,Hangup() exten => b2c-sip,1,NoOp() same => n,Read(Digits,outgoing_welcome,0,,5,500) same => n,Set(GLOBAL(DIGITS)=${Digits});Before starting a business initiated call, add customer WA header to store the WA user number captured from agent entered digits (DTMF) same => n,Dial(PJSIP/1000,,b(handler^addheader^1)) same => n,Hangup() exten => c2b-sip,1,NoOp() same => n,Read(Digits,incoming_welcome,0,,5,500) same => n,Dial(PJSIP/${Digits}) same => n,Hangup()pjsip.confReplace the following placeholders with actual values {external-media-address}: Public IP of the Asterisk server for media{external-signaling-address}: Public IP of the Asterisk server for signaling{local-net}: local network of the Asterisk server{sip-ua-password}: Chosen SIP User Agent passwordNote:Extension 1000 is used to bridge WA calls with Asterisk see section Integration with Cloud API Signaling [global] type=global debug=yes ;Enable/Disable SIP debug logging.Valid options include yes|no[transport-tcp] type=transport protocol=tcp bind=0.0.0.0;External IP address to usein RTP handling external_media_address={external-media-address};External address for SIP signalling external_signaling_address={external-signaling-address};Network to consider local used for NAT purposes local_net={local-net}[endpointtemplate](!) type=endpoint context=default disallow=all allow=OPUS,g722,g729,ulaw ;No audio if direct_media isset to yes direct_media=no rtp_symmetric=yes use_avpf=yes media_encryption=sdes media_use_received_transport=yes rtcp_mux=yes [authtemplate](!) type=auth auth_type=userpass password={sip-ua-password}[aortemplate](!) type=aor max_contacts=1 remove_existing=yes [1000](endpointtemplate) disallow=all ;extension 1000is used byRtpEngine to bridge whatsapp calls ;WhatsApp only support OPUS allow=OPUS auth=1000_auth aors=1000[1000_auth](authtemplate) username=1000[1000](aortemplate)[1001](endpointtemplate) auth=1001_auth aors=1001[1001_auth](authtemplate) username=1001[1001](aortemplate)[1002](endpointtemplate) auth=1002_auth aors=1002[1002_auth](authtemplate) username=1002[1002](aortemplate)[1003](endpointtemplate) auth=1003_auth aors=1003[1003_auth](authtemplate) username=1003[1003](aortemplate)[1004](endpointtemplate) auth=1004_auth aors=1004[1004_auth](authtemplate) username=1004[1004](aortemplate)[1005](endpointtemplate) auth=1005_auth aors=1005[1005_auth](authtemplate) username=1005[1005](aortemplate) Configuring a VoIP phoneRefer to the earlier section Final checklist Double-check all configuration files for correct numbers, passwords, and domain names.Make sure your firewall allows SIP (5060/TCP) and RTP (10000-20000) ports.For more details on SIP password setup, see the WhatsApp Cloud API documentation. Troubleshooting Cannot register SIP UAConfirm that the SIP URL is correct and the domain is pointing to the Asterisk server. Run host {domain-name} to verify that the IP address points to the Asterisk server. Asterisk with built-in WebRTC using Graph APIThis approach is similar to Asterisk using Graph API with RtpEngine except that it uses the built-in WebRTC support in Asterisk and hence does not require RtpEngine.The RtpEngineClient component is hence not required in this approachIn terms of configuration and setup, only difference is the configuration of extension 1000 which is given below ...;Rest of content omitted for brevity [1000](endpointtemplate) disallow=all ;extension 1000is used by SIP UA of the integration module to bridge WhatsApp calls ;WhatsApp only support OPUS allow=OPUS auth=1000_auth aors=1000 dtls_auto_generate_cert=yes webrtc=yes ;Setting webrtc=yes is a shortcut for setting the following options:; use_avpf=yes ; media_encryption=dtls ; dtls_verify=fingerprint ; dtls_setup=actpass ; ice_support=yes ; media_use_received_transport=yes ; rtcp_mux=yes