# Technical Capabilities

We provide detailed overview of the platform’s extensive technical capabilities, including support for various protocols such as SIP and RTP, as well as audio codecs.

### Protocols

|   Protocol   | Description                                                                                                                                                                                                     |
| :----------: | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
|      SIP     | SIP signaling, as specified in [IETF RFC 3261](https://datatracker.ietf.org/doc/html/rfc3261).                                                                                                                  |
|      TLS     | The Transport Layer Security (TLS) is a cryptographic protocol that provides secure communication over a network. Voice Gateway supports TLS 1.2 and later versions. TLS 1.3 is preferred for optimal security. |
| SIP over TLS | Partially encrypted SIP, as specified in [IETF RFC 3261](https://datatracker.ietf.org/doc/html/rfc3261). Some servers may communicate unencrypted, depending on specific requirements.                          |
|     SIPS     | Fully encrypted SIP, as specified in [IETF RFC 3261](https://datatracker.ietf.org/doc/html/rfc3261). All servers in the communication chain use TLS for encrypted communication.                                |
|      RTP     | The Real-time Transport Protocol as specified in [RFC 3550](https://datatracker.ietf.org/doc/html/rfc3550).                                                                                                     |
|     SRTP     | Encrypted media using the Secure Real-time Transport Protocol (SRTP), as specified in [RFC 3711](https://datatracker.ietf.org/doc/html/rfc3711).                                                                |
|     DTMF     | The use of RTP payloads to carry DMTF events, as specified in [RFC 2833](https://datatracker.ietf.org/doc/html/rfc2833).                                                                                        |
|   SIP REFER  | Sending the SIP REFER method to transfer calls, as specified in [RFC 3515](https://datatracker.ietf.org/doc/html/rfc3515). Receiving SIP REFER is not supported.                                                |

### Codecs

* `G.722` native.
* `G.711`&#x20;
* `G.729`
* `OPUS` (only SBC).

### Capabilities

Trunk management and Routing

* Transferring calls via [SIP REFER](https://datatracker.ietf.org/doc/html/rfc3515) or [SIP INVITE](https://datatracker.ietf.org/doc/html/rfc3261).
* Multiple SIP trunks per customer.
* Configuring SIP trunks with options like tech prefix, SIP Diversion header, Outbound authentication (including `REGISTER`).
* Routing calls based on a trunk group, Direct Inward Dialing (DID), or DID range.
* Least-cost routing selection of outbound trunk.

### Features

* Custom SIP headers on inbound and outbound calls.
* Mid-call SIP INFO requests.
* P-Asserted-Identity header to identify caller.
* Receiving compact SIP headers.
* Receiving re-INVITE with no Session Description Protocol (SDP).
* Configurable music on hold.


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