When it comes to the secure transmission of multimedia data in IP networks, they are THE decisive factor: security protocols that ensure encrypted data transfer. For tap-proof conversations in voice-over-IP telephony (VoIP), you should rely on SRTP.
But what is SRTP? How does encryption with SRTP work? And what is the difference between SRTP and RTP? Our article clarifies.
1. What is RTP?
RTP is the short form for Real-Time Transport Protocol. It is used to transmit multimedia content in real time over the IP network. RTP is classically used in the area of streaming and communication, such as video conferencing or telephone systems.
RTP uses UDP (User Datagram Protocol), which works connection-free. Data is transmitted unsecured with this protocol. Data stream control is handled by the Real-Time Control Protocol – RTCP for short.
2. How RTP and SRTP work
SRTP is the extension of RTP: The abbreviation SRTP stands for Secure Real-Time Transport Protocol and refers to the encrypted real-time transmission of data. While the transmission with RTP is not tap-proof and can be recorded, with SRTP all audio data is encoded and transmitted as encrypted packets, so that privacy, message authentication and playback protection are guaranteed. The following graphic shows a comparison of RTP and SRTP:
The RTP packet identifies the media payload type – the data format – and its source. The RTP header contains, among other things, version and sequence numbers, sender ID, timestamp, and synchronization information. This header is unencrypted. This also applies to the user data – the RTP payload.
In the SRTP packet, however, the RTP header is authenticated and the payload is encrypted. The recipient can decrypt it using a master key.
3. Encryption of VoIP calls
SRTP is ideally suited for VoIP telephony because it ensures secure voice encryption without affecting voice quality. Authentication and encryption close the security gaps of RTP. Data is encrypted as soon as a call is established. In order to hear the respective conversation partner, the data packets must be decrypted on receiver side by using a master key.
To ensure that calls via the internet are completely secure, it is not enough to encrypt only the media data. For a safe connection, the metadata of the respective session must also be encrypted. This is done via SIP or SIPS – the Session Initiation Protocol Secure. The SIP is responsible for call control and handles the establishment and termination of the connection between the participants. SIPS is the extension of the protocol and contains the encryption protocol TLS (Transport Layer Security) – formerly SSL – which enables the secure establishment of internet connections. In the area of voice over IP, SIPS is used to establish an encrypted connection between the telephone system and the IP telephone, for example.
4. Secure IP telephony with TENIOS
So, for tap-proof telephony on the internet, users should rely on telecom providers that use both SRTP and SIPS.
Just like TENIOS:
With telephony solutions from TENIOS, all call data is optimally protected at all times. Use our PBX solution or connect your existing IP-capable telephone system to the telephone network via SIP trunk.
What to expect from TENIOS:
- Individual telephony solutions with flexible scalability
- Extensive functions and features without additional payment
- Hosting in our geo-redundant data centers in Germany
- Highest data and failover security
- Intuitive & self-explanatory user portal
- Extensive documentation & support
- No minimum term, can be cancelled monthly
- Attractive prices in tour pay-as-you-go model