SIP provider configuration information and questionnaire for BYOC Cloud/ BYOC Premises trunks
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.
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
- 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
You can tell your SIP provider that these RFCs are supported by Genesys Cloud.
|RFC 791||Internet Protocol (IPv4)|
|RFC 2327||SDP: Session Description Protocol|
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 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 Servers
Note: 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|
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)|
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.
You can then use the answers to configure each setting for the trunk configurations within Genesys Cloud. For more information, see Create a trunk under BYOC Premises.