SIP provider configuration information and questionnaire for BYOC Cloud/ BYOC Premises trunks

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Genesys Cloud requires specific IP address support, SIP OPTIONS requests, out-of-band DTMF, fax pass-through, inbound/outbound calls with ANI, early media, hold support, 503 responses, and sequential failover from the SIP provider. Genesys Cloud supports SIP and RTP traffic over ports 5060 and UDP/TCP and RFCs related to networking, signaling, media, and security. The provider should allow traffic from Genesys Cloud IP addresses and use G.711 codec while partially supporting RFC 3263 and avoiding crypto key changes for SRTP. Genesys Cloud SIP trunking connects your telephony system to the Genesys Cloud voice platform, allowing you to leverage call routing, interactive voice response, call recording, and analytics while using your existing infrastructure. Configuring SIP trunks involves discussing options with your provider and validating functionality. SIP trunking enables Genesys Cloud's voice features while integrating with on-premises telephony.

When working with your SIP provider to set up and configure your BYOC (Premises or Cloud) external trunk in Genesys Cloud, you need to provide them with information about Genesys Cloud. You also need to ask specific questions about configuration options. Once you have your initial trunk configuration in place, you want to run some tests to make sure that everything is set up properly.

In this article, we provide you with a list of Genesys Cloud requirements. We also provide you with a list of the RFCs currently supported in Genesys Cloud. You can use this information when working with your SIP provider.

Also in this article, we provide you with a questionnaire designed to help you discuss your needs and requirements with your SIP provider. We also provide you with a test plan designed to help you make sure that your SIP trunk is configured properly.

RequirementsClick to expand

Discuss these requirements with your SIP provider.

List IP addresses the carrier will use to source SIP and RTP traffic
SIP OPTIONS request to each Edge device for trunk failover

Functionality requirements

Out-of-Band DTMF (RFC2833)
Fax via g.711 pass-though
Inbound calls with and without ANI
Outbound calls with ANI
Early Media
Hold Support (re-invite to 0.0.0.0)
Incoming call forwarding
Always in Audio
Service unavailable (503 response from carrier)

SIP/RTP traffic requirements

Carrier must allow SIP and RTP traffic from your Edge Appliance IP Addresses
Sequential trunk failover is recommended for multi-Edge deployments
SIP port 5060 for all inbound/outbound SIP traffic
Recommended to use the G.711 Codec for voice traffic inbound and outbound between you and the carrier

RFCs supported by Genesys CloudClick to expand

You can tell your SIP provider that these RFCs are supported by Genesys Cloud.

Standard	Description
RFC 791	Internet Protocol (IPv4)
RFC 2327	SDP: Session Description Protocol
RFC 2833	RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals The customer endpoints should be able to handle in-band if RFC 2833 is not supported (by the PBX and/or the far end).
RFC 4733	RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals. The method used to transmit dual-tone multifrequency (DTMF) signaling. Set to RTP Events for RFC 2833 or 4733 compliancy.
RFC 3261	SIP: Session Initiation Protocol The transport methods supported are UDP (RFC 768) and TCP (RFC 793). The PBX-specific agreed transport method (i.e. UDP or TCP) shall apply in both directions between the PBX and the service.
RFC 3263	Session Initiation Protocol (SIP): Locating SIP Servers Note: Genesys Cloud only partially supports this RFC as SRV and NAPTR are excluded.
RFC 3264	An Offer/Answer Model with Session Description Protocol (SDP)
RFC 3311	The Session Initiation Protocol (SIP) UPDATE Method For more information see, Does Genesys Cloud support RFC 3311?
RFC 3515	The Session Initiation Protocol (SIP) Refer Method
RFC 3550	RTP: A Transport Protocol for Real-Time Applications
RFC 3711	The Secure Real-time Transport Protocol (SRTP) > Transform-independent parameters. Note: Genesys changes the crypto key on every re-INVITE. As such, it is crucial that carriers honor the new crypto key to avoid issues with one-way audio.
RFC 3960	Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP),
RFC 4028	Session Timers in the Session Initiation Protocol (SIP) For more information see, Does Genesys Cloud support RFC 4028?
RFC 4244	An Extension to the Session Initiation Protocol (SIP) for Request History Information
RFC 4566	SDP: Session Description Protocol (obsoletes RFC 2327)
RFC 4904	Representing Trunk Groups in tel/sip Uniform Resource Identifiers (URIs)
RFC 5806	Diversion Indication in SIP (network accepts does not send)
ITU G.168	Echo cancellation
RFC 5923	Connection Reuse in the Session Initiation Protocol (SIP)

QuestionnaireClick to expand

Download the questionnaire, which is a fillable PDF form, before you contact your SIP provider. Then use it to discuss the SIP trunk configuration options with your SIP provider and record the answers.

Download our SIP trunk questionnaire

You can then use the answers to configure each setting for the trunk configurations within Genesys Cloud. For more information, seeCreate a trunk under BYOC Premises.

Test planClick to expand

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SIP provider configuration information and questionnaire for BYOC Cloud/ BYOC Premises trunks

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