Work with your carrier to provision an BYOC Premises external SIP trunk

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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.
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   RequirementsClick to expand

   
   

   • 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 currently supported in Genesys CloudClick to expand

   
   

   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)

   
   
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.