Secure Communication with SRTP and key managemnt protocols like SDES , ZRTP and DTLS

With advent of Voice over IP , the real time streaming of data/audio/video also became critically important to be protected from eavesdropping or modification over the open internet.

While Secure Real-time Transport Protocol (SRTP) is a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP).

ZRTP is a protocol that negotiates the keys and other information required to setup a SRTP audio and video session

To read about RealTime Transport protocol (RTP) , RTP control protocol (RTCP ) , before reading about adding security to RTP , RTCP and its feedback go here .

SRTP (Secure Real-time Transport Protocol)

SRTP provides a framework for encryption and message authentication of RTP and RTCP streams by negotiating keys.

It is not a transport but a profile of the Real-time Transport Protocol (RTP) for securing RTP streams in addition to providing confidentiality, integrity protection, source authentication, and replay protection.

The SRTP specification also defines how to setup and maintain a cryptographic context. This context holds all necessary data to perform the security operations, for example the SRTP encryption keys, the packet sequence counters, authentication keys, and so on. Each SRTP session, which is the same as a RTP session, has its own context. Thus a bidirectional SRTP communication requires two different SRTP cryptographic contexts.

Features of SRTP

framework for encryption and message authentication of RTP and RTCP streams
confidentiality and integrity of the entire RTP and RTCP packets, together with protection against replayed packets.
secure for unicast and multicast RTP applications
low computational cost and small footprint
high throughput and low packet expansion to support bandwidth economy.
permits upgrading with new cryptographic transforms,
protection for heterogeneous environments (mix of wired and wireless networks)

Independant from the underlying transport, network, and physical layers used by RTP, in particular high tolerance to packet loss and re-ordering.

Normal RTP Packet
SecureRTP Packet

SRTCP ( Secure RTCP)

Secure RTCP (SRTCP) is similar to the SRTP format of the SRTCP packet which has the authentication tag and MKI headers, including two additional headers:

  • SRTCP index
  • Encrypt-flag

Key management protocols for SRTP

Since SRTP does not contain an integrated key management solution, one can employ any of the following key management protocols

SDES (Session Description Protocol Security Descriptions) – SRTP Key management

It is a way to negotiate the key/cryptographic parameters for SRTP.
Keys are transported in the SDP attachment of a SIP message using TLS transport layer (SSLv3/TLSv1) or other methods like S/MIME.

media attribute defined by SDES is “crypto”
a=crypto: inline: [session-parms]

SDES packet

3 commonly used crypto suites are :

  1. AES_CM_128_HMAC_SHA1_80
  2. AES_CM_128_HMAC_SHA1_32
  3. F8_128_HMAC_SHA1_32

DTLS – SRTP Key management

DTLS keying happens on the media path, independent of any out-of-band signalling channel present.

Jitsi Client SRTP configuration

An offer can include any of –

  • plain RTP (RTP/AVP),
  • RTP with RTCP-based feedback (RTP/AVPF),
  • Secure RTP (RTP/SAVP), or
  • Secure RTP with RTCP-based feedback (RTP/SAVPF)

SDP for RTP/AVP

v=0
o=987654321-jitsi.org 0 0 IN IP4 x.x.x.x.
s=-
c=IN IP4 x.x.x.x
t=0 0
m=audio 24380 RTP/AVP 9
a=rtcp-xr:voip-metrics
a=rtpmap:9 G722/8000
a=sendrecv
m=audio 24400 RTP/AVP 9
a=rtcp-xr:voip-metrics
a=rtpmap:9 G722/8000
a=sendrecv

or

v=0.
o=987654321-jitsi.org 0 0 IN IP4 x.x.x.x.
s=-.
c=IN IP4 x.x.x.x.
t=0 0.
m=audio 5018 UDP/TLS/RTP/SAVP 9.
a=rtpmap:9 G722/8000.
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level.
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level.
a=rtcp-xr:voip-metrics.
a=setup:actpass.
a=fingerprint:sha-1 B9:0F:89:EE:BD:1F:B1:C4:86:B6:D7:5C:25:88:53:F4:02:F4:F5:91.
m=audio 5018 RTP/SAVPF 9.
a=rtpmap:9 G722/8000.
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level.
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level.
a=rtcp-xr:voip-metrics.
a=setup:actpass.
a=fingerprint:sha-1 B9:0F:89:EE:BD:1F:B1:C4:86:B6:D7:5C:25:88:53:F4:02:F4:F5:91.

The m line indicates which mode of RTP and RTCP is it offering .

Case where offerer/calleer wants to establish a Secure RTP audio stream on plain RTP with DTLS-SRTP as the key management protocol.

type: offer, sdp: 
v=0
o=- 2977074634695769063 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=msid-semantic: WMS i2CKXQdort5QF76tyO5SUKyyyyPfMYR4kjZO
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:w5/T
a=ice-pwd:zuPM49QcEX3cKRQiKylJU4Y6
a=ice-options:trickle
a=fingerprint:sha-256 5A:70:05:55:C1:5A:82:51:02:D3:00:A3:BF:E7:EF:62:DF:29:EB:F2:9F:5F:51:58:12:D9:4C:AA:41:36:86:13
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:i2CKXQdort5QF76tyO5SUKyyyyPfMYR4kjZO 5ffdb0f9-48b1-43bc-9f63-ea032643aeba
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2215726670 cname:e6egqLfRbLu6vH45
a=ssrc:2215726670 msid:i2CKXQdort5QF76tyO5SUKyyyyPfMYR4kjZO 5ffdb0f9-48b1-43bc-9f63-ea032643aeba
a=ssrc:2215726670 mslabel:i2CKXQdort5QF76tyO5SUKyyyyPfMYR4kjZO
a=ssrc:2215726670 label:5ffdb0f9-48b1-43bc-9f63-ea032643aeba
m=application 9 DTLS/SCTP 5000
c=IN IP4 0.0.0.0
a=ice-ufrag:w5/T
a=ice-pwd:zuPM49QcEX3cKRQiKylJU4Y6
a=ice-options:trickle
a=fingerprint:sha-256 5A:70:05:55:C1:5A:82:51:02:D3:00:A3:BF:E7:EF:62:DF:29:EB:F2:9F:5F:51:58:12:D9:4C:AA:41:36:86:13
a=setup:actpass
a=mid:2
a=sctpmap:5000 webrtc-datachannel 1024

SRTP on kamailio

For Secure Communication kamailio supports – Digest SIP User authentication , Authorization via ACL or group membership , IP and Network authentication , TLS support for SIP signaling , transparent handling of SRTP for secure audio , TLS domain name extension support ,authentication and authorization against database (MySQL, PostgreSQL, UnixODBC, BerkeleyDB, Oracle, text files), RADIUS and DIAMETER.

Code to set flag rtp_secure_media to true if both TLS and SRTP are active

<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">	<action application="set" data="rtp_secure_media=true"/></condition>

Invite from Jitsi client alternatively offering 3 different types of audio SDP’s – RTP/SAVPF , RTP/SAVP and RTP/AVP. Which ever will be accepted by the other endpoint will be communicated back using SDP in 200 OK.

INVITE sip:99999999999@x.x.x.x:5080 SIP/2.0
   Call-ID: 2a34d1e981602c82c345513f3f2f89ed@0:0:0:0:0:0:0:0
   CSeq: 1 INVITE
   From: "altanai" ;tag=bed49270
   To: 
   Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK-3130-9657d2ae9b662779bc08cdd32881828f
   Max-Forwards: 70
   Contact: "altanai" 
   User-Agent: Jitsi2.10.5550Mac OS X
   Content-Type: application/sdp
   Content-Length: 2336
   v=0
   o=7777777777-jitsi.org 0 0 IN IP4 y.y.y.y
   s=-
   c=IN IP4 y.y.y.y
   t=0 0
   m=audio 5016 UDP/TLS/RTP/SAVP 9
   a=rtpmap:9 G722/8000
   a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=rtcp-xr:voip-metrics
   a=setup:actpass
   a=fingerprint:sha-1 55:CF:25:5D:D5:65:71:C8:D9:FF:97:AD:CC:F2:08:DB:38:DD:81:38
m=audio 5016 RTP/SAVPF 9
   a=rtpmap:9 G722/8000
   a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=rtcp-xr:voip-metrics
   a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Ekb2qAA8F7VCmz0FMSrad0rIt8duHQFedu/KxMbD
   a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:rEeGiaLCUbFw0sS0FxARgX9i5pwEj/frxxbgGkch
   a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:up9VO2T/rfu8V0cecA4RuG0aWgSaCC5gD/p/RdY1odg1p/0Pto0=
   a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:6yLDM31gAuwrlL0qkH72QYJLwtzX1IX+Z+7UML3VA5CpIbUWeAw=
   a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:2Q3b3UpPJMosXTrm/0Ui5q3Mw8tQ6ig5Xq0jt4Ibj0t5hVQx5KBRbC+8sMJDMg==
   a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:yVs8C3xPFY2LAUXIH+dlgBBNSz+jm1cbAQlAgv8hPKGe1zfu2wzx1d465UfFzQ==
   a=crypto:7 F8_128_HMAC_SHA1_80 inline:bhIPhj1TryAB63p/g8B3gL5NXJJ7V4kbjXqYaU54
   a=setup:actpass
   a=fingerprint:sha-1 55:CF:25:5D:D5:65:71:C8:D9:FF:97:AD:CC:F2:08:DB:38:DD:81:38
m=audio 5016 RTP/SAVP 9
   a=rtpmap:9 G722/8000
   a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=rtcp-xr:voip-metrics
   a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Ekb2qAA8F7VCmz0FMSrad0rIt8duHQFedu/KxMbD
   a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:rEeGiaLCUbFw0sS0FxARgX9i5pwEj/frxxbgGkch
   a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:up9VO2T/rfu8V0cecA4RuG0aWgSaCC5gD/p/RdY1odg1p/0Pto0=
   a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:6yLDM31gAuwrlL0qkH72QYJLwtzX1IX+Z+7UML3VA5CpIbUWeAw=
   a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:2Q3b3UpPJMosXTrm/0Ui5q3Mw8tQ6ig5Xq0jt4Ibj0t5hVQx5KBRbC+8sMJDMg==
   a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:yVs8C3xPFY2LAUXIH+dlgBBNSz+jm1cbAQlAgv8hPKGe1zfu2wzx1d465UfFzQ==
   a=crypto:7 F8_128_HMAC_SHA1_80 inline:bhIPhj1TryAB63p/g8B3gL5NXJJ7V4kbjXqYaU54
m=audio 5016 RTP/AVP 9
a=rtpmap:9 G722/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics

Kamailio in secure mode selects the SRTP block of Audio SDP and responds in 200 OK

RTP to SRTP Bridging in Freeswitch

Enable ZRTP globally. Can override this on a per channel basis http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)

When SRTP it’s critical to not offer or accept variable bit rate codecs, doing so would leak information and possibly compromising SRTP stream. (FS-6404)

Supported SRTP Crypto Suites:

AEAD_AES_256_GCM_8

This algorithm is identical to AEAD_AES_256_GCM (see Section 5.2 of [RFC5116]), except that the tag length, t, is 8, and an
authentication tag with a length of 8 octets (64 bits) is used. An AEAD_AES_256_GCM_8 ciphertext is exactly 8 octets longer than its
corresponding plaintext.

AEAD_AES_128_GCM_8

This algorithm is identical to AEAD_AES_128_GCM (see Section 5.1 of [RFC5116]), except that the tag length, t, is 8, and an
authentication tag with a length of 8 octets (64 bits) is used. An AEAD_AES_128_GCM_8 ciphertext is exactly 8 octets longer than its
corresponding plaintext.

AES_CM_256_HMAC_SHA1_80 | AES_CM_192_HMAC_SHA1_80 | AES_CM_128_HMAC_SHA1_80

AES_CM_128_HMAC_SHA1_80 is the SRTP default AES Counter Mode cipher and HMAC-SHA1 message authentication with an 80-bit authentication
tag. The master-key length is 128 bits and has a default lifetime of a maximum of 2^48 SRTP packets or 2^31 SRTCP packets, whichever comes
first.

AES_CM_256_HMAC_SHA1_32 | AES_CM_192_HMAC_SHA1_32 | AES_CM_128_HMAC_SHA1_32

This crypto-suite is identical to AES_CM_128_HMAC_SHA1_80 except that the authentication tag is 32 bits. The length of the base64-decoded key and salt value for this crypto-suite MUST be 30 octets i.e., 240 bits; otherwise, the crypto attribute is considered invalid.

AES_CM_128_NULL_AUTH

The SRTP default cipher (AES-128 Counter Mode), but to use no authentication method. This policy is NOT RECOMMENDED unless it is unavoidable; see Section 7.5 of [RFC3711].

SRTP variables that modify behaviors based on direction/leg:

rtp_secure_media

possible values:
mandatory – Accept/Offer SAVP negotiation ONLY
optional – Accept/Offer SAVP/AVP with SAVP preferred
forbidden – More useful for inbound to deny SAVP negotiation
false – implies forbidden
true – implies mandatory

default if not set is accept SAVP inbound if offered.

rtp_secure_media_inbound | rtp_secure_media_outbound

This is the same as rtp_secure_media, but would apply to either inbound
or outbound offers specifically.

How to specify crypto suites:

By default without specifying any crypto suites FreeSWITCH will offer crypto suites from strongest to weakest accepting the strongest each
endpoint has in common. If you wish to force specific crypto suites you can do so by appending the suites in a comma separated list in the order that you wish to offer them in.

Examples:
rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80

Additionally you can narrow this down on either inbound or outbound by specifying as so:

rtp_secure_media_inbound=true:AEAD_AES_256_GCM_8
rtp_secure_media_inbound=mandatory:AEAD_AES_256_GCM_8
rtp_secure_media_outbound=true:AEAD_AES_128_GCM_8
rtp_secure_media_outbound=optional:AEAD_AES_128_GCM_8

rtp_secure_media_suites

Optionaly you can use rtp_secure_media_suites to dictate the suite list and only use rtp_secure_media=[optional|mandatory|false|true] without having to dictate the suite list with the rtp_secure_media* variables.

In vars.xml

SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls valid options: sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2 default: tlsv1,tlsv1.1,tlsv1.2

TLS cipher suite: default ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH The actual ciphers supported will change per platform. openssl ciphers -v ‘ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH’ Will show you what is available in your verion of openssl.

SRTP to RTP over multiple Crypto suits

Logs :

A client at 7777777777@ is trying to call 9999999999@ , which freeswtch has to proxy and convert from RTP to SRTP.
The following debug logs form sofia external show this process.

recv 1215 bytes from udp/[]:4642 at 07:08:27.374857:


INVITE sip:9999999999@:5080;transport=UDP SIP/2.0
   Via: SIP/2.0/UDP :47851;branch=z9hG4bK-524287-1---7cc8ad9383e9787d;rport
   Max-Forwards: 70
   Contact: :47851;transport=UDP>
   To: :5080;transport=UDP>
   From: :5080;transport=UDP>;tag=5df9f82c
   Call-ID: lFNvnuABQfOpROxfFp-MZQ..
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
   Content-Type: application/sdp
   User-Agent: Z 5.2.28 rv2.8.115
   Allow-Events: presence, kpml, talk
   Content-Length: 607
v=0
   o=Z 20472192 0 IN IP4 
   s=Z
   c=IN IP4 
   t=0 0
   m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
   a=rtpmap:106 opus/48000/2
   a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
   a=rtpmap:111 speex/16000
   a=rtpmap:97 iLBC/8000
   a=fmtp:97 mode=20
   a=rtpmap:110 speex/8000
   a=rtpmap:112 speex/32000
   a=rtpmap:98 telephone-event/48000
   a=fmtp:98 0-16
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:100 telephone-event/16000
   a=fmtp:100 0-16
   a=rtpmap:99 telephone-event/32000
   a=fmtp:99 0-16
   a=rtpmap:102 G726-32/8000
   a=sendrecv

[NOTICE] switch_channel.c:1104 New Channel sofia/external/7777777777@:5080 [ed5e07ee-bd00-4a47-b4e1-6abc9dd23ed6]

[DEBUG] switch_core_state_machine.c:584 (sofia/external/7777777777@:5080) Running State Change CS_NEW (Cur 1 Tot 33)

[DEBUG] sofia.c:10078 sofia/external/7777777777@:5080 receiving invite from :4642 version: 1.9.0 -742-8f1be0 64bit

[DEBUG] sofia.c:7291 Channel sofia/external/7777777777@:5080 entering state [received][100]

[DEBUG] sofia.c:7301 Remote SDP:
v=0
o=Z 20472192 0 IN IP4 
s=Z
c=IN IP4 
t=0 0
m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=rtpmap:102 G726-32/8000
[DEBUG] sofia.c:7693 (sofia/external/7777777777@:5080) State Change CS_NEW -> CS_INIT
State NEW
Running State Change CS_INIT (Cur 1 Tot 33)
Standard INIT
State Change CS_INIT -> CS_ROUTING
State INIT going to sleep
Running State Change CS_ROUTING (Cur 1 Tot 33)
Callstate Change DOWN -> RINGING
State ROUTING

send 389 bytes to udp/[]:4642 at 07:08:27.376085:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP :47851;branch=z9hG4bK-524287-1---7cc8ad9383e9787d;rport=4642;received=
   From: :5080;transport=UDP>;tag=5df9f82c
   To: :5080;transport=UDP>
   Call-ID: lFNvnuABQfOpROxfFp-MZQ..
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit
   Content-Length: 0

After the invote is recived and processed with 100 trying reply , the routing and rtp secure trabformation begins by adding crypto keys and forarding to

Standard EXECUTE
ed5e07ee EXECUTE sofia/external/7777777777@:5080 set(rtp_secure_media=optional)
[rtp_secure_media]=[optional]
ed5e07ee EXECUTE sofia/external/7777777777@:5080 log(INFO Forwarding calls 9999999999@ )
Forwarding calls 9999999999@
…
Set Local audio crypto Key [1 AEAD_AES_256_GCM_8 inline:aHJ1yquBtm4Lzfi2oMpe6cV7IBEy3YgKxrJ3qjvLuRXSuZfHcV4VtVNwHDw]
Set Local video crypto Key [1 AEAD_AES_256_GCM_8 inline:qeJbqlSbnKBNew575hSZ3LX78o6GBsjgOrSMxzGH/zb1E7mkls1Mda93U9w]
Set Local text crypto Key [1 AEAD_AES_256_GCM_8 inline:VghMVsjWQwnOAAjBJ1NTB3jZgfpNV/Yu4poxkAPMqkC7C+fhPKApCJrWg3U]
Set Local audio crypto Key [2 AEAD_AES_128_GCM_8 inline:7XNrjjwC/eOVnWlBSp74DfiIGAEYn/BN+latfA]
Set Local video crypto Key [2 AEAD_AES_128_GCM_8 inline:UQrFpy9Q7L5DI/ww4e5IAmwy7BxSw5yd/T0v0Q]
Set Local text crypto Key [2 AEAD_AES_128_GCM_8 inline:ZqkEPrUFHkaQ+7CROp52H/JO0MbrYWk/Eyl9lQ]
Set Local audio crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:PTGAm2KlbfuKtIUVGtXknKKzALAzfILZJuPOjfO9S07eWRE6FR0aMUvjuehJgw]
Set Local video crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:ahHIB0o/dp3SliYWK9BkxM7TfzILwG0bjDn7JuvYi+puRkTM4mYvvsSmywLaYA]
Set Local text crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:crAs8dPcWJkEEGj5nqTvFGl/TWpxxb86k+dX5gBXhh+q6DO2pEqWNkQmm55aLA]
Set Local audio crypto Key [4 AES_CM_192_HMAC_SHA1_80 inline:SLBJWjgMdfiYX7TUwWQ9CmqUsILLJrpBIVjbfuQmpBIFLvvA/XU]
Set Local video crypto Key [4 AES_CM_192_HMAC_SHA1_80 fNazWgWwNRPjUKNHVqkz44]
Set Local text crypto Key [4 AES_CM_192_HMAC_SHA1_80 inline:hbe9qqETBSK5hRQ8DI9mXL4QAjjGSR8tGDiTHCJF3yxCrRk1ajk]
Set Local audio crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:8q8mer9N2V4qVxnaazuJeT0KXgW2scONy36J3KaS]
Set Local video crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:TP5NQ1yB8ZSCCwZMgXur9VHZ5SlpNfnXePj7eZrk]
Set Local text crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:HT3F3iYG8H/majhBZbOs2Z8ye/WEVGT5Oytx2oQS]
Set Local audio crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:fEohh92lX2xLmeFYlt8YouM2jN4z5pU05d90BYfoAKU6m4CWv8g8AnifDUKk9A]
Set Local video crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:+uBNmLcvj41hXoMxNlMNBpq68gU4PmLwYcdopEB/X/jfPElkUgHfguPIgIFJUg]
Set Local text crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:cqk7D3+KMQ+31R4FFDRRzn/aluyIgjxBL59vfxcsdf5OW9izEJtU+06GewJyIA]
Set Local audio crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:Tv25TfP9fQZ+ljs/tFlHohkckiK4F6cemzEjHSvo2+q6No4ai+o]
Set Local video crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:CY/Dizd1QrlobZtgnigr0hWE+oDSx4S1F51Zpo4aZamN+8ZMdp8]
Set Local text crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:aEox/7IMps5c+uOWbosZ618+opkJV/GnrKc2EnAhVnDNeo91+No]
Set Local audio crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:0LwKGyljIed0zhukiMMyD5ive0ZsyybwBrnevcAv]
Set Local video crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:eZN8rAG8UPPntdYxsg1kkWL4qMsVgTiGGiS4UeUM]
Set Local text crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:bAYzbfr+El8usaTkPBR6iFuTda4uLNGjyx9lQWkX]
Set Local audio crypto Key [9 AES_CM_128_NULL_AUTH inline:5m3142gGG1HZ5VnoXsAOyopSwDCYbrIsGpdbEO3D]
Set Local video crypto Key [9 AES_CM_128_NULL_AUTH inline:zXk67wjwRhSilq0kiz5TWxXqrxuTaWTA3qqbVo/G]
Set Local text crypto Key [9 AES_CM_128_NULL_AUTH inline:FRP9CJbBO+PRj6I9RSBAiMxRZ/qFtyrEXPfxocG0]
sending invite version: 1.9.0 -742-8f1b7e0 64bit
Local SDP:
v=0
o=FreeSWITCH 1552960557 1552960558 IN IP4
s=FreeSWITCH
c=IN IP4
t=0 0
m=audio 18750 RTP/SAVP 102 9 0 8 103 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 telephone-event/48000
a=fmtp:103 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AEAD_AES_256_GCM_8 inline:aHJ1yquBtm4Lzfi2oMpe6cV7IBEy3YgKxrJ3qjvLuRXSuZfHcV4VtVNwHDw
a=crypto:2 AEAD_AES_128_GCM_8 inline:7XNrjjwC/eOVnWlBSp74DfiIGAEYn/BN+latfA
a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:PTGAm2KlbfuKtIUVGtXknKKzALAzfILZJuPOjfO9S07eWRE6FR0aMUvjuehJgw
a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:SLBJWjgMdfiYX7TUwWQ9CmqUsILLJrpBIVjbfuQmpBIFLvvA/XU
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:8q8mer9N2V4qVxnaazuJeT0KXgW2scONy36J3KaS
a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:fEohh92lX2xLmeFYlt8YouM2jN4z5pU05d90BYfoAKU6m4CWv8g8AnifDUKk9A
a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:Tv25TfP9fQZ+ljs/tFlHohkckiK4F6cemzEjHSvo2+q6No4ai+o
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:0LwKGyljIed0zhukiMMyD5ive0ZsyybwBrnevcAv
a=crypto:9 AES_CM_128_NULL_AUTH inline:5m3142gGG1HZ5VnoXsAOyopSwDCYbrIsGpdbEO3D
a=ptime:20
a=sendrecv

Once the SDP is ready with crypto keys it is the forwarded to the next_up

send 2104 bytes to udp/[]:5060 at 07:08:27.378167:


INVITE sip:9999999999@ SIP/2.0
   Via: SIP/2.0/UDP :5080;rport;branch=z9hG4bKmF251mK2pN35B
   Max-Forwards: 69
   From: "7777777777" >;tag=vcKeKD6SN02cB
   To: >
   Call-ID: a27898fd-c4b8-1237-ddaa-02a933b32da0
   CSeq: 1935861 INVITE
   Contact: :5080>
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 1304
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "7777777777" >;party=calling;screen=yes;privacy=off
v=0
   o=FreeSWITCH 1552960557 1552960558 IN IP4 
   s=FreeSWITCH
   c=IN IP4 
   t=0 0
   m=audio 18750 RTP/SAVP 102 9 0 8 103 101
   a=rtpmap:102 opus/48000/2
   a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:103 telephone-event/48000
   a=fmtp:103 0-16
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AEAD_AES_256_GCM_8 inline:aHJ1yquBtm4Lzfi2oMpe6cV7IBEy3YgKxrJ3qjvLuRXSuZfHcV4VtVNwHDw
   a=crypto:2 AEAD_AES_128_GCM_8 inline:7XNrjjwC/eOVnWlBSp74DfiIGAEYn/BN+latfA
   a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:PTGAm2KlbfuKtIUVGtXknKKzALAzfILZJuPOjfO9S07eWRE6FR0aMUvjuehJgw
   a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:SLBJWjgMdfiYX7TUwWQ9CmqUsILLJrpBIVjbfuQmpBIFLvvA/XU
   a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:8q8mer9N2V4qVxnaazuJeT0KXgW2scONy36J3KaS
   a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:fEohh92lX2xLmeFYlt8YouM2jN4z5pU05d90BYfoAKU6m4CWv8g8AnifDUKk9A
   a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:Tv25TfP9fQZ+ljs/tFlHohkckiK4F6cemzEjHSvo2+q6No4ai+o
   a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:0LwKGyljIed0zhukiMMyD5ive0ZsyybwBrnevcAv
   a=crypto:9 AES_CM_128_NULL_AUTH inline:5m3142gGG1HZ5VnoXsAOyopSwDCYbrIsGpdbEO3D
   a=ptime:20

Multimedia Internet Keying (MIKEY) – Key management of SRTP

can establish multiple security contexts or cryptographic sessions with a single message.
Can be used in p2p or bradcast scenarios where one entity generates the key and needs to distribute the key to a number of participants

modes of operatiosn

  • Pre-Shared Key
  • Public Key Encryption
  • Diffie-Hellman
  • HMAC-Authenticated Diffie-Hellman
  • RSA-R
  • TICKET
  • IBAKE
  • SAKKE

References


RFC 3711 The Secure Real-time Transport Protocol (SRTP)
RFC 3611 RTP Control Protocol Extended Reports (RTCP XR)

SRTP in Freeswitch – https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/libs/srtp


WebRTC Security

Unlike most conventional  real-time systems (e.g., SIP-based soft phones) WebRTC communications  are directly controlled by a Web server over some signalling protocol which may be XMPP , websockets , socket.io , Ajax etc . This poses new challenges such as

  • Web browser might expose a JavaScript APIs which allows  web server to place a video call itself.This may cause web pages to secretly record and stream the webcam activity from user’s computer
  • malicious calling services can record the user’s conversation and misuse
  • malicious webpages can lure users via advertising and execute auto calling services .
  • Since JavaScript calling APIs are implemented as browser built-ins , un authorized access to these can also make user’s audio and camera streams vulnerable
  • If program and APIs allow the server to instruct the browser to send arbitrary content, then they can be used to bypass firewalls or mount denial of service attacks.

The general goal of security is to identify and resolve security issues during the design phase so they do not cost service provider time, money, and reputation at a later phase. Security for a large architecture project involves many aspects, there is no one device or methodology to guarantee that an architecture is now “secure” Areas that malicious individuals will attempt to attack include but are not limited to:

  • Improperly coded applications
  • Incorrectly implemented protocols
  • Operating System bugs
  • Social engineering and phishing attacks

As security is a broad topic touching on many sections of WebRTC this section is not meant to address all topics but instead to focus on specific “hot spots”, areas that require special attention due to the unique properties of the WebRTC service. There are several security-related topics that are of particular interest with respect to WebRTC.  They can be grouped into the following areas:

  1. Identity Management
  2. Browser Security
  3. Authentication
  4. Media encryption

The are discussed in detail below :

Identity Management

Support of WebRTC should not increase security risk to telecom network. Any device or software that is in the hands of the customer will be compromised, it is just a mater of time

  • All data received from untrusted sources (i.e. all data from customer controlled devices or software) must be validated.
  • Any data sent to the client will be obtained by malicious users

Provide exceptional protection for our customer’s data and make all reasonable attempts at protecting the customer from their own mistakes that may compromise their own systems. —Ensure that the new service does not adversely impact the data security, privacy, or service of existing customers.

Browser Security

Specific security concerns include:

Cross-site scripting (XSS)

A type vulnerability typically found in Web applications (such as web browsers through breaches of browser security) that enables attackers to inject client-side script into Web pages viewed by other users.

  • A cross-site scripting vulnerability may be used by attackers to bypass access controls such as the same origin policy.
  • Cross-site scripting carried out on websites accounted for roughly 80.5% of all security vulnerabilities documented by Symantec as of 2007 according to Wikipedia.
  • Their effect may range from a petty nuisance to a significant security risk, depending on the sensitivity of the data handled by the vulnerable site and the nature of any security mitigation implemented by the site’s owner.

As the primary method for accessing WebRTC is expected to be using HTML5 enabled browsers there are specific security considerations concerning their use such as; protecting keys and sensitive data from cross-site scripting or cross-domain attacks, websocket use, iframe security, and other issues. —Because the client software will be controlled by the user and because the browser does not, in most cases, run in a protected environment there are additional chances that the WebRTC client will become compromised. This means all data sent to the client could be exposed.

  • keys
  • hashes
  • registration elements (PUID etc.)

Therefore additional care needs to be taken when considering what information is sent to the client, and additional scrutiny needs to be performed on any data coming from the client.

Clickjacking

(User Interface redress attack, UI redress attack, UI redressing) is a malicious technique of tricking a Web user into clicking on something different to what the user perceives they are clicking on, thus potentially revealing confidential information or taking control of their computer while clicking on seemingly innocuous web pages. It is a browser security issue that is a vulnerability across a variety of browsers and platforms, a clickjack takes the form of embedded code or a script that can execute without the user’s knowledge, such as clicking on a button that appears to perform another function. —Compromised personal computer with installed adware, viruses, spyware such as trojan horses, etc. can also compromise the browser and obtain anything the browser sees.

Authentication

Authentication happens on different levels

End user Authentication

Through UID ( unique ID ) of USER

Device Authentication

  • SIM enabled devices follow standard IMS-AKA authentication
  • Non-SIM enabled “devices” are authenticated using user authentication

Application Authentication

  • Model mirrors current application onboarding procedures.
  • Application developers need to establish service agreement
  • Client_Id secrets are exchanged as part of this process.
  • Use  security gateway for authenticating applications

The Browser Threat Model

The browser acts as a TRUSTED COMPUTING BASE (TCB) both from the user’s perspective and to some extent from the  server’s.  HTML and JavaScript (JS) provided by the web server can execute scripts on browser and generate actions and events . However browser  operates in a sandbox that isolates these scripts both from the user’s computer and from server .

Access to Local Resources

The users computer may have lot of private and confidential data on the disk . Browser do make it mandatory that user must explicitly select the file and consent to its upload before doing file upload and transfer transactions . However still it is not very rare that misleading text and buttons can make users click files .  

Another way of accessing local resources is through downloading malicious files to users computer which are executable and may harm users computer.

SOP or Same Origin Policy

SOP  forces scripts from each site to run in their own, isolated, sandboxes.  It enables webpages and scripts from the same origin server to interact with each other’s JS variables, but prevents pages from the different origins or even iframes on the same page to not exchange information.

As part of SOP scripts are allowed to make HTTP requests via the  XMLHttpRequest() API to only those server which have same ORIGIN/domain as that of the originator .

CORS [Cross-Origin Resource Sharing ]

CORS enables multiple web services to intercommunicate . Therefore when a script from origin A executes what would otherwise be a forbidden cross-origin request, the browser instead contacts the target server B to determine whether it is willing to allow cross-origin requests from A.  If it is so willing, the browser then allows the request.  This consent verification process is designed to safely allow cross-origin requests.

Websockets

Even websockets overcome SOP and establish cross origin transport channels .

Once a WebSockets connection has been established from a script to a site, the script can exchange any traffic it likes without being required to frame it as a series of HTTP request/response transactions.

WebSockets use masking technique to randomize the bits that are being transmitted , thus making it more difficult to generate traffic which resembles a given protocol , thus making it difficult for inspection from flowing traffic .

JSONP

Jsonp is a hack designed to bypass origin restriction through script tag injection. A JSONp enabled server passes the response in user specified function

when we use <script> tags the domain limitation is ignored ie we can load scripts from any domain .  So when we need to fetch get exchange data just pass callback parameters through scripts . For example

function mycallback(data){
// this is the callback function executed when script returns
alert("hi"+ data);</span>
}
var script = document.createElement('script');
script.src = '//serverb.com/v1/getdata?callback=mycallback'
document.head.appendChild(script) 

There have been found vulnerabilities in the existing Java and Flash consent verification techniques and handshake.

Security around ICE and TURN

ICE

Sender and receiver are able to share media stream after a offer answer handshake. But we already need one in order to do NAT hole-punching. Presuming the ICE server is malicious , in absence of transaction IDs by stun unknow to call scripts , it is not possible for the webpage of receiver to ascertain is the data is forged or original . Thus to prevent this the browser must generate hidden transaction Id’s and should not sharing with call scripts ,even via a diagnostic interface.

IP Location Privacy

As soon as the callee sends their ICE candidates, the caller learns the callee’s IP addresses.  The callee’s server reflexive address reveals a lot of information about the callee’s location.

To prevent server should suppress the start of ICE negotiation until the callee has answered.

Also user may hide their location entirely by forcing all traffic through a TURN server.

Communications Security

Goal of webrtc based call services should be to create channel which is secure  against both message recovery and message modification for all audio / video and data .

Threats from Screen Sharing

With the increasing requirement of screen sharing in web app and communication systems there is always a high threat of oversharing / exposing confidential passwords , pins , security details etc . This may either through some part of screen or some notification whihc pops up .

There is always the case when the user may believe he is sharing a window when in fact they are the entire desktop.

The attacker may request screensharing and make user open his webmail , payment settings or even net-banking accounts .

Long term access to camera and microphone

When user frequently uses a site he / she may want to give the site a long-term access to the camera and microphone ( indicated by ” Always allow on this site ” in chrome ). However the site may be hacked and thus initiate call on users’ computer automatically to secretly listen-in .

False UI shows cut off call while still being active

Unless the user checks his laptops glowing camera light LED or goes and monitors the traffic himself he would not know if there is active call in background, which according to him he had cut off . In such a case an attacker may pretend to cut a call shows red phone signs and supportive text but still keep the session and media stream active placing himself on mute .

During-Call Attack

Even if the calling service cannot directly access keying material ,it  can simply mount a man-in-the-middle attack on the connection. The idea is to mount a bridge capturing all the traffic.

To protect against this it is now mandatory to use https for using getusermedia and otherwise also recommended to keep webrtc comm services on https or use strict fingerprinting .
This section is derived from Security Considerations for WebRTC draft-ietf-rtcweb-security-08

So does existing WebRTC model offer security ?

We know that the forces behind WebRTC standardization are WHATWG, W3C, IETF and strong internet working groups . WebRTC security was already taken into consideration when standards were being build for it . The encryption methods and technologies like DTLS and SRTP were included to safeguard users from intrusions so that the information stays protected.

WebRTC media stack has native built-in features that address security concerns. The peer-to-peer media is already encrypted for privacy . Figure below:

WebRTC media stack Solution Architecture - Google Slides (1)
WebRTC media stack

Media Encryption

Primary issue with supporting DTLS is it can put a heavy load on the SBC’s handling encryption/decryption duties. —Interworking DTLS-SRTP to SDES is CPU intensive

  • SRTP from DTLS-SRTP end flows easily
  • SRTP from SDESC end requires auth+decrypt, and encrypt+auth

Reason:  DTLS-SRTP handshake has both ends choose “half” of the SRTP key The Encrypted Key Transport (EKT) proposed by Cisco solves this problem and provides additional security. Recommendation is to use DTLS-SRTP with EKT enhancements . Note: In order to avoid potential security issues, the SRTP authentication tag length used by the base authentication method must be at least ten octets.

For WebRTC to transfer real time data, the data is first encrypted using the DTLS (Datagram Transport Layer Security) method. This is a protocol built into all the WebRTC supported browsers from the start (Chrome, Firefox and Opera). On a DTLS encrypted connection, eavesdropping and information tampering cannot take place.

Other than DTLS, WebRTC also encrypts video and audio data via the SRTP (Secure Real-Time Protocol) method ensuring that IP communications – your voice and video traffic – can not be heard or seen by unauthorized parties.

What is SRTP ?

The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications.

Earlier models of VOIP communication such as SIP based calls had an option to use only RTP for communication thereby subjecting the endpoint users to lot of problem like compromising media Confidentiality  . However the WebRTC model mandates the use of SRTP hence ruling out insecurities of RTP completely. For encryption and decryption of the data flow SRTP utilizes the Advanced Encryption Standard (AES) as the default cipher.

What is DTLS ?

DTLS allows datagram-based applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. The DTLS protocol is based on the stream-oriented Transport Layer Security (TLS) protocol .

Together DTLS and SRTP enables the exchange of the cryptographic parameters so that the key exchange takes place in the media plane and are multiplexed on the same ports as the media itself without the need to reveal crypto keys in the SDP.

Today the browser acts as a TRUSTED COMPUTING BASE (TCB) where the HTML and JS act inside of a sandbox that isolates them both from the user’s computer.

A script cannot access user’s webcam , microphone , location , file , desktop capture without user’s explicit consent. When the user allows access, a red dot will appear on that tab, providing a clear indication to the user, that the tab has media access.

Figure depicting browser asking for user’s consent to access Media devices for WebRTC .

Untitled drawing

Figure depicting Media Capture active on browser with red dot .

Untitled drawing (1)
we know that XMLHttpRequest() API can be used to secretly send data from one origin to other and this can be used to secretly send information without user’s knowledge . However now , SAME ORIGIN POLICY (SOP) in browser’s prevents server A from mounting attacks on server B via the user’s browser, which protects both the user (e.g., from misuse of his credentials) and the server B (e.g., from DoS attack).

Security challenges with Web Server based WebRTC service are many for example :

  1. If the both the peers have WebRTC browser then one can place a WebRTC call to callee anytime this might result in denial of service .
  2. Since the media is p2p and also can override firewalls settings through TURN server , it can result in unwanted data being send to peer .
  3. One may secretly make calls to users through website and extract information .
  4. Threat from screen sharing, for example user might mistakenly share his internet banking screen or some confidential information.
  5. Giving long-term access to the camera and microphone for certain sites is also a concern . for example : since next time you visit a site that has access to your microphone and camera , they can secretly be viewing youe webcam and microphone inputs .
  6. Clever use of User Interface to mask a ongoing call can mislead the user into believing that call has been cut while it is secretly still ongoing.
  7. Network attackers can modify an HTTP connection through my Wifi router or hotspot to inject an IFRAME (or a redirect) and then forge the response to initiate a call to himself.
  8. As WebRTC doesn’t yet have an congestion control mechanism , it can eat up a large chunk of user’s bandwidth.
  9. By visiting chrome://webrtc-internals/ in chrome browser alone , one can view the full traces of all webRTC communication happening through his browser . The traces contain all kinds of details like signalling server used , relay servers , TURN servers , peer IP , frame rates etc .
WebRTC Internals

Ofcourse other challenges that arrive with any other webservice based architecture are also applicable here such as :

  1. Malicious Websites which automatically execute the attacker’s scripts.
  2. User can be induced to download harmful executable files and run them.
  3. Improper use of W3C Cross-Origin Resource Sharing (CORS) to bypass SAME ORIGIN POLICY (SOP) .

How can I make my WebRTC solution secure ?

In the recent months everyone has been trying to get into the WebRTC space but at the same time fearing that hackers might be able to listen in on conferences, access user data, or even private networks. Although development and usage around WebRTC is so simple , the security and encryption aspects of it are in the dim light.

Best practices to make your VOIP Solution more secure

A simple WebRTC architecture is shown in the figure below :

WebRTC media stack Solution Architecture - Google Slides (2)

By following the simple steps described below one can ensure a more secure WebRTC implementation . The same applies to healthcare and banking firms looking forth to use WebRTC as a communication solution for their portals .

Ensure that the signalling platform is over a secure protocol such as SIP / HTTPS / WSS . Also since media is p2p , the media contents like audio video channel are between peers directly in full duplex.

To protect against Man-In-The-Middle (MITM) attack the media path should be monitored regularly for no suspicious relay.

User’s that can participate in a call , should be pre registered / Authenticated with a registrar service. Unauthenticated entities should be kept away from session’s reach .

WebRTC authentication certificate
WebRTC authentication certificate

Make sure that ICE values are masked thereby not rendering the caller/ callee’s IP and location to each other through tracing in chrome://webrtc-internals/ or packet detection in Wirehsark on user’s end.

As the signalling server maintains the number of peers , it should be consistently monitored for addition of suspicious peers in a call session. If the number of peers actually present on signalling server is more that the number of peers interacting on WebRTC page then it means that someone is eavesdropping secretly and should be terminated from session access by force.

It was observed that manyatimes non tech savy users simply agree to all permissions request from browser without actually consciously giving consent . Therefore user’s should be made aware of API in websites which ask for undue permissions . For example permission to :

Screenshot from 2015-04-22 15:22:15

Third party API should be thoroughly verified before sending their data on WebRTC DataChannel.

Before Desktop Sharing user’s should be properly notified and advised to close any screen containing sensitive information .

Keystore and master key protection

For storing logs , recording , file , ssh keys or any others ensitive informaton encrypted by keys , we need a safe storage for keys and these tools are handy for password and key management – Dashlane , Lastpass , Bitwarden, 1Password

Auto sign-in security measure for WebRTC apps

Turn User Authentication On and enable Two-Factor Authentication/Bio-metrics.

OTP based signons and captcha checks are also popular approaches to protect signins

Avoid Public Wi-Fi

Even a WebRTC connection can be tampered with an insecure Wifi . Even if the Man-in-middle cannot decipher message content he can make out the packet size , frequency , end parties in siagnling , time delay etc . Also as a precautionary measure enable Remote Lock and Data Wipe and only use authorized apps with permission to sensitive data such as image storage.

Native WebRTC apps

If you use a native WebRTC native app , there are mulitple thinsg that you need to be wary of.

Avoid All Jailbreaks : Jail-breaking a smartphone can enable the user to run unverified or unsupported apps, many of these apps carry security vulnerabilities. Majority of security exploits for Apple’s iOS only affect jailbroken iPhones.

Add a Mobile Security App : Mobile security reports shows that mobile operating systems such as iOS and (especially) Android are increasingly becoming targets for malware. Select a reputable mobile security app that extends the built-in security features of the device’s mobile operating system. Some well-known third-party security vendors offering mobile security apps for iOS, Android and Windows Phone – Avast , Kaspersky , Symantec

Also as a good practise Turn off the Bluetooth, Wi-Fi and NFC when not needed.

Information security

Forum : Huawei

Information security ensures that both physical and digital data is protected from unauthorized access, use, disclosure, disruption, modification, inspection, recording or destruction. 

Intentional Security breaches

Although WebRTC already has best secure tools in its spec list which provide end to end encrypted communication over SRTP DTLS as well as media device access mandatory from websites of secure origin over TLS, yet if the endpoints acting as peers themselves are compromised then all this is in vain . Hence security issues arises when

  • the endpoints are recording their media content and storing it on unsafe location such as public file servers or
  • endpoints are inturn re-streaming their incoming their media streams to unsafe streaming servers

Exploiting human vulnerabilities / Unintentional breaches

Phishing , Pretexting , Baiting attacks , Quid pro quo , Tailgating , Water-Holing are soe of the common tactics to steal teh data of a nonsuspecting user . They are as much applicate to WebRTC based communication site as they are to any other trusted website such as banking , customer care contacts , falsh sale portals , cupon / discount sites etc .

Phone phishing – Voice phishing a criminal phone fraud, imporsonating legitimate caller such as a bank or tax agent and using social engineering over the telephone system to gain access to private personal and financial information for the purpose of financial reward.

SMS phishing – SMS is used as a method to do phishing maya tiem to send malicious links posing as legitmate sender

Other social engineering tactics – Trickery , Influencing , Deception , Spying

Impersonation attacks – spear-phishing , emails that attempt to impersonate a trusted individual or a company in an attempt to gain access to corporate finances , Human resource details , sesitive data. Business email compromise (BECs) also known as CEO fraud is a popular example of an impersonation attack. The fake email usually describes a very urgent situation to minimize scrutiny and skepticism.

Network security breaches

Inspite of the fact that webrtc is a p2p streaming framework , there are always signalling server required which do the initial handshake and enable the exchange fo SDP for the media to stream in peer to peer fashion . Some wellknown attacks that compromise networks and remote / cloud server are :

  • Viruses, worms and Trojan horses
  • Zero-day attacks
  • Hacker attacks
  • Denial of service attacks
  • Spyware and adware

It is upto the WebRTC/ VoIP service provider to detect emerging threats before they infiltrate network and compromise data. Some crticial compoenets to enhance security are Firewalls , Access Control Lists , Intrusion detection and prevention systems (IDS/IPS) , Virtual private networks (VPN)

Governance Framework – defines the roles, responsibilities and accountability of each person and ensures that you are meeting compliance.

  • Confidentiality: ensures information is inaccessible to unauthorized people via encryption
  • Integrity: protects information and systems from being modified by unauthorized people; provides accuracy and trustworthyness
  • Availability: ensures authorized people can access the information when needed and that all hardware and software are maintained properly and updated when necessary
  • Authentication, Authorization and Accountability(AAA): 
    validate users autheticity via creds, enforcing policies on network resources after the user has gain access and ensuring accountability by means of monitoring and capturing the events done by the user
  • Non repudiation: is the assurance that someone cannot deny the validity of something. It provides proof of the origin of data and the integrity of the data.

What happens if your VOIP solution is on the verge of being compromised ?

As a first defence tactic , if a orignation ip address is sending malacious or malformed packets which depict an exploitation or attack , trigger and notification for tech team and execute script to block on the origin IP of attacker via security groups in AWS or other ACL list in hosted server . Can also implement temporary firewall block on it and later monitor it for more violations.

Incase a server is compromised beyond repair such as attacker taking control of the file system , drain the ongoing sessions from it and store cached storage with session state variable like CDR enteries . Activate the fallback / standby server and make the current server a honeypot to explore the attackers actions

Commonly attacks involve either

  • exploiting the VoIP system to get free internatoinal calls
  • ransomware activities such as scp the files out of server and leaveing behind a readme.txt file on root location asking for money transfer in return of data
  • bombard brute force DDOS attacks to bring down the system and make it incapible of catering to genuine requests , perhaps with the inetention of giving advantage to competitors .

As the media connections are p2p , even if we kill the signalling server , it will not affect the ongoing media sessions . Only the time duration ( probably 3 – 4 minutes ) it takes to restart the server , is when the users will not be able to connect to signalling server for creating new sessions . therefore incase a system is under attack and non recoverable , just terminate it and respawn other server attaced to the domain name or floating IP or Load balancer.

Most browsers today like Google Chrome and Mozilla Firefox have a good record of auto-updating themselves withing 24 hours of a vulnerability of threat occurring .

If a call is confirmed to be compromised , it should be within the power of Web Application server rendering the WebRTC capable page to cut off the compromised call session by force censing termation request to endpoints or via turning off the TURN services if in use .

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