Secure, encrypted VoIP service

Transport Layer Security (TLS) is a cryptographic protocol that provides message security. As it relates to SIP, TLS protects SIP messages sent by your PBX or softphone with encryption. SIP headers contain your called and calling numbers and other authentication data. With TLS properly configured, these messages become impossible for the attacker or eavesdropper in the middle, to read or modify.

In addition to message security, with TLS turned on, VoIP can be unlocked on many mobile networks around the world that normally block SIP. Switching to TLS also solves some of the NAT issues inherent to internet access over 3G/LTE. In many non-deregulated, restrictive jurisdictions such as the Gulf states, use of TLS over non-standard port can unlock SIP calling in a heavily censored environment.

The optional Secure Real-time Transport Protocol (SRTP) was developed by a team of cryptographic experts from Cisco and Ericsson, and published by the IETF in 2004. SRTP encrypts your RTP audio streams between your on-site IP PBX or Unified Communication platform and the regional DID Logic gateway.

Use of SRTP on the trunk ensures that no third parties are able to decrypt, modify or tamper with the audio stream during the call.

You can choose to only use TLS without SRTP (as in some Lync/Skype for Business setups when adding external SIP providers), or to use TLS and SRTP. FreeSWITCH supports SRTP via SDES. Asterisk 1.8 has native support for SRTP. Cisco CUBE, 3CX, Sonus, Genband and Avaya have their own implementations and can be configured to support SRTP.

In addition to SRTP clients can use a dedicated cross-connect at participating data centers such as Telecity AMS04 and One Wilshire. If you’re colocating your PBX with a hosting provider or running virtual PBX instances, check out SwitchPorts – your free, zero config private link straight into our network over private gigabit Cisco switching fabric.