Work with your carrier to provision your BYOC Premises external SIP trunk. There are four major steps:

  1. Provide the Genesys Cloud requirements and list of supported RFCs to your carrier.

    • 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
    • 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

    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
    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: It's important the carrier doesn't change the crypto key.

    RFC 3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP),
    RFC 4028 Session Timers in the Session Initiation Protocol (SIP)
    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)


  2. Work with your carrier to fill out the BYOC Premises SIP trunk questionnaire with the required technical information.
  3. Use the Create a trunk under BYOC Premises procedure to change the external trunk configuration settings in Genesys Cloud to match the carrier’s answers to the SIP trunk questionnaire.
  4. Complete the SIP trunk test plan.