Work with your carrier to provision an BYOC Premises external SIP trunk
Work with your carrier to provision your BYOC Premises external SIP trunk. There are four major steps:
- 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
- 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
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).
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.
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.
Session Initiation Protocol (SIP): Locating SIP ServersNote: Genesys Cloud only partially supports this RFC as SRV and NAPTR are excluded.
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 parametersNote: 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)
- Work with your carrier to fill out the BYOC Premises SIP trunk questionnaire with the required technical information.
- 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.
- Complete the SIP trunk test plan.