SIP methods

tyntec acts as a gateway, so tyntec only transports SIP and RTP data.

The following SIP methods are supported:




Indicates a client is being invited to participate in a call session. tyntec supports privacy values like CLIR, according to RFC 3325. In case of a forwarded call, the diversion key will be filled with a value.


Confirms that the client has received a final response to an INVITE request.


Terminates a call and can be sent by either the caller or the recipient.


Cancels any pending request.


Sends mid-session information that does not modify the session state. Can be used for DTMF information.


Queries the capabilities of servers. Request available only for load balancing feature.


Other requests, such as, but not limited to, REGISTER and UPDATE, are not supported.

The typical call flow between tyntec and the customer is shown in the standard call diagram in the Connection Steps section for Terminating Voice Calls and Receiving Voice Calls. This diagram is an example of a standard scenario that will apply to most call flows on the platform; more complex scenarios, like 183 Session progress answers, are possible as well.

tyntec uses port 5060 for all SIP messages. For RTP traffic, tyntec uses a port range from 35000 to 65000. SIP and RTP use Datagram (UDP) as the transport layer protocol.

All addresses must be public IP addresses. Hostname resolving and fully qualified domain names (FQDNs) are not supported. In case the customer is connected to tyntec over a VPN, the IP addresses are not public.