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Secure Media

Secure Media uses encryption to ensure that the call media and associated signaling remains private during transmission. Transport Layer Security (TLS) provides encryption for SIP signaling. Secure Real-time Transport Protocol (SRTP) provides encryption for call content/media packets.

SRTP provides a framework for the encryption of RTP & RTCP. RFC 4568, Session Description Protocol (SDP) Security Description (SDES) for Media Streams, defines such a protocol specifically designed to exchange cryptographic material using a newly defined SDP crypto attribute.

Inbound

You can enable or disable Secure Media in your SIP Domain. It is disabled by default.

You can expect the following:

Enabled: TLS must be used to encrypt SIP messages and SRTP must be used for the media packets. Any non-encrypted calls will be rejected.
Disabled: RTP must be used for media packets. SIP messages may be sent in the clear or using TLS. Any SRTP encrypted calls will be rejected.

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Info

SRTP supports the following crypto suites: AES_CM_128_HMAC_SHA1_80 and AES_CM_128_HMAC_SHA1_32. Both may be included in an order of preference.
The optional master key identifier (MKI) parameter is not supported

Outbound

Ensure you configure secure=true parameter as part of SIP URI to secure media in SIP outbound calls.

1<?xml version="1.0" encoding="UTF-8"?>
2<Response>
3  <Dial>
4    <Sip>sip:jack@example.com;secure=true</Sip>
5  </Dial>
6</Response>

The default port 5061 will be used for TLS.

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Info

Only a single crypto suite for SRTP will be included: AES_CM_128_HMAC_SHA1_80
The optional master key identifier (MKI) parameter is not supported

TLS/SRTP support with Asterisk

Asterisk ships by default with chan_sip driver and works well with Twilio. However, if you have some reason to run PJSIP driver with Asterisk, please note the following:

Here is a guide to installing a non-bundled version of PJSIP. Change the version to 2.5.5 in the steps.

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Warning

Asterisk 13.8 cert2 defaults to PJSIP 2.5 which will not work with Twilio for TLS/SRTP purposes. Non-encrypted calls will still work.

Make sure to use the latest PJSIP driver, which at this time is 2.5.5.

You may see following message in your log:

ERROR[10886]: pjproject:0 <?>: tlsc0x7f217c03 RFC 5922 (section 7.2) does not allow TLS wildcard certificates. Advise your SIP provider, please!

This message can be ignored.