Attacks on SIP Networks

Major standards bodies including 3GPP, ITU-T, and ETSI have all adopted SIP as the core signalling protocol for services such as LTE , VoIP, conferencing, Video on Demand (VoD), IPTV (Internet Television), presence, and Instant Messaging (IM) etc. With the continous evolution of SIP as the defacto VoIP protocol , we need to underatdn the risk mitigartion practices around it .

I have written about VoIP and security in these blogs before

For Security around web browser based calling via webrtc i have written

  • Webrtc Security , which describes browser threat modal , access to local resource , Same Orogin Policy (SOP) and Cross Resource Sharing ( CORS) as well as Location sharing , ICE , TUEN and threats to privacy with screen sharing , microgone camera long term access and probable mid call attacks .
  • Genric secrutity of web Application build around hosting platform of webrtc. Includs concepts like Identity management , browser security – cross site security amd clickjacking , Authetication of devices and applications , Media Encryption and regex checking.

Also Written about VoIP security at protocol level with SRTP /DTLS using TLS and specifically using available pre added security modules on kamailio SIP server. It describes Sanity checks , ACL lists with permissions , hiding topology details , countering Flood using pike and Fail2Ban as well as Traffic monitoring and detection .

In this article we will cover types of attacks on SIP systems

Types of attacks on SIP based systems

Registration Hijacking

malicious registrations on registrar by a third party who modifies From header field of a SIP request.

exmaple implementation :
attacker de-registers all existing contacts for a URI
attacker can also register their own device as the appropriate contact address, thereby directing all requests for the affected user to him

solution – Autheticaion of user

Impersonating a Server

attacker impersonates the remote server
user’s request can now be intercepted by some other party
user’s request may be forwarded to insecure locations

Solution –

confidentiality, integrity, and authentication of proxy servers

Proxy/redirect sever, and registrars SHOULD possess a site certificate issued by CA which could be validated by UA

Temparing Message bodies

If users are relying on SIP message bodies to communicate either of

  • session encryption keys for a media session
  • MIME bodies
  • SDP
  • encapsulated telephony signals
    Then the atackers on proxy server can modify the session key or can act as a man-in-the-middle and do eaves droppng

exmaple implementation :
attacker can point RTP media streams to a wiretapping device
can changes Subject header field to appear to users as spam

solution – end to end ecryption over TLS + Digest Authorization

mid-session threats like tearing down session

Request forging
attacker learns the params of the session like To , From tags etc then he can alter ongoing session parameters and even bring it down

example implementation :
attacker inserts a BYE in a ongoing session thereby tearing it down
can insert re INVITE and redierct the stream to wiretaping device

solution – authetication on every request
signing and encrypting of MIME bodies, and transference of credentials with S/MIME

Denial of Service and Amplification

DOS attacks – rendering a particular network element unavailable, usually by directing an excessive amount of network traffic at its interfaces.
dDOS – multiple network hosts to flood a target host with a large amount of network traffic.

Can be created by sending falsified sip requests to other parties such that numerous transactions originating in the backwards direction comes to the target server created congestion.

exmaple implementation :
attackers creates a falsified source IP address and a corresponding Via header field that identify a targeted host as the originator of the request. Then send this to large number of SIP network element . This geneerates DOS aimed at target.

attackers uses falsified Route header field values in a request that identify the target host and then send such messages to forking proxies that will amplify messaging sent to the target.

Flooding with register attacks can deplete available memory and disk resources of a registrar by registering huge numbers of bindings.
Flooding a stateful proxy server causes it to consume computational expense associated with processing a SIP transaction

Solution –
detect flooding and pike in traffic and use ipban to block
challenge questionable requests with only a single 401 (Unauthorized) or 407 (Proxy Authentication Required), forgoing the normal response retransmission algorithm, and thus behaving statelessly towards unauthenticated requests.

Security mchanisms

Full encryption vs hop by hop encrption

SIP mssages cannot be encrypted end-to-end in their entirety since
message fields such as the Request-URI, Route, and Via need to be visible to proxies in most network architectures
so that SIP requests are routed correctly.
proxy servers need to also update the message with via headers

Thus SIP uses low level security along with hop by hop encrption and auth headers to verify the identity of proxy servers

Transport and Network Layer Security

IPsec – used where set of hosts or administrative domains have an existing trust relationship with one another.

TLS – used where hop-by-hop security is required between hosts with no pre-existing trust association.

SIPS URI Scheme

Used as an address-of-record for a particular user, signifies that each hop over which the request is forwarded, must be secured with TLS

HTTP Authentication

Reuse of the HTTP Digest authentication via 401 and 407 response codes that implement challenge for autehtication
provides replay protection and one-way authentication.

S/MIME

allows SIP UAs to encrypt MIME bodies within SIP, securing these bodies end-to-end without affecting message headers.
provides end-to-end confidentiality and integrity for message bodies

nonce-count

provides replay protection

SIP over TLS

SIP messages can be secured using TLS. There is also TLS for Datagrams called DTLS.

Security of SIP signalling is different from security of protocols used in concert with SIP like RTP , RTCP. and that will be covered in later topics of this article.

TLS operation consists of two phases: handshake phase and bulk data encryption phase

Handshake phase

Prepare algorithm to be used during TLS session

Server Authentication

server sends its certificate to the client, which then verifies the certificate using a certificate authority’s (CA’s) public key.

Client Authentication

Server sends an additional CertificateRequest message to request the client’s certificate. The client responds with

  1. Certificate message containing the client certificate with the client public key and
  2. CertificateVerify message containing a digest signature of the handshake messages signed by clients private key

Server authenticates client by client’s public key , since only client holding correct private key can sign the message.

prepare the shared secret for bulk data encryption

client generate a pre_master_secret, and encrypt it using the server’s public key obtained from the server’s certificate. The server decrypts the pre_master_secret using its own private key.
Both the server and client then compute a master_secret they share based on the same pre_master_secret. The master_secret is further used to generate the shared symmetric keys for bulk data encryption and message authentication

Public key cryptographic operations such as RSA are much more expensive than shared key cryptography. This is why TLS uses public key cryptography to establish the shared secret key in the handshake phase, and then uses symmetric key cryptography with the negotiated shared secret as the data encryption key.

Stateless proxy servers do not maintain state information about the SIP session and therefore tend to be more scalable. However, many standard application functionalities, such as authentication, authorization, accounting, and call forking require the proxy server to operate in a stateful
mode by keeping different levels of session state information.

Steps :

  1. The SIP proxy server enforces proxy authentication with
    407 Proxy Authentication Required challenge.
  2. UAC provides credentials that verify its claimed identity (e.g., based on MD5 [34] digest algorithm) and retransmits in authorization header

Security of RTP

confidentiality protection of the RTP session and integrity protection of the RTP/RTCP packets requires source authentication of all the packets to ensure no man-in-the-middle (MITM) attack is taking place.

end to end media encryption – SRTP ( Secure RTP )

encodes the voice into encrypted IP packages and transport those via the internet from the transmitter  to receive

References

  • The Impact of TLS on SIP Server Performance – Charles Shen† Erich Nahum‡ Henning Schulzrinne† Charles Wright , Department of Computer Science, Columbia University,IBM T.J. Watson Research Center

Certificates, compliances and Security in VoIP

This article describes various Certificates and compliances, Bill and Acts on data privacy, Security and prevention of Robocalls as adopted by countries around the world pertaining to Interconnected VoIP providers, telecommunications services, wireless telephone companies etc

Compliance certificates by Industry types

HIPAA (Health Insurance Portability and Accountability Act)

Deals with privacy and security of personal medical records and electronic health care transaction

Applicability  : If voip company handles medical information

Includes : 

  • Not allowed Voice mail transcription
  • Should have End-to-End Encryption
  • Restrict  using unsecured WiFi networks to prevent Snooping
  • User security , strong password rules  and mandatory monthly change
  • Secure Firmware on VoIP phones
  • Maintaining Call and Access Logs

SOX( Sarbanes Oxley Act of 2002)

Also known as SOX, SarbOX or Public Company Accounting Reform and Investor Protection Act

Applicability : if managing the communications operations of a regulated, publicly traded company 

Includes : 

  • Retain records which include financial and other sensitive data
  • ways employees are provided or denied access to records or data based on their roles and responsibilities
  • do information audit by a trusted third party. 
  • Retention and deletion of files such as audio files like voicemails, text messages, video clips, declared paper records, storage, and logs of communications activities
  • Physical and digital security controls around cloud-based VoIP applications and the networks

Privacy Related Compliance certificates

COPPA (Children’s Online Privacy Protection Act ) of 1998 

prohibits deceptive marketing to children under the age of 13, or collecting personal information without disclosure to their parents. 

any information is to be passed on to a third party, must be easy for the child’s guardian to review and/or protect

2011 amendment  requires that the data collected was erased after a period of time,

2014 FTC issued guidelines that apps and app stores require “verifiable parental consent.”

CPNI (Customer Proprietary Network Information) 2007

CPNI (Customer Proprietary Network Information) in united states is the information that communication providers  acquire about their subscribers. This Individually identifiable information that is created by a customer’s relationship with a provider, such as data about the frequency, duration, and timing of calls, the information on a customer’s bill, and call identifying information. This processing information is governed strictly by FCC and certification should be renewed on an annual basis

Provider can pass along that information to marketers to sell other services, as long as the customer is notified

In 2007, the FCC explicitly extended the application of the Commission’s CPNI rules of the Telecommunications Act of 1996 to providers of interconnected VoIP service.

CALEA

Communications Assistance for Law Enforcement Act (CALEA) conduct electronic surveillance by imposing specific obligations on “telecommunications carriers” for assisting law enforcement, including delivering call interception and call identification functionality to the government with a minimum of interference to customer service and privacy.

Read more about CALEA and its roles in VoIP here Regulatory and Legal Considerations with WebRTC development

GDPR (General Data Protection Regulation)  in European Union 2018

Supersedes the 1995 Data Protection Directive

Establishes requirements of organizations that process data, defines the rights of individuals to manage their data, and outlines penalties for those who violate these rights.

No personal data may be processed unless this processing is done under one of six lawful bases specified by the regulation (consent, contract, public task, vital interest, legitimate interest or legal requirement). When the processing is based on consent the data subject has the right to revoke it at any time.

Controllers must notify Supervising Authorities (SA)s of a personal data breach within 72 hours of learning of the breach.

California Consumer Privacy Act (CCPA) 2019

consumer rights relating to the access to, deletion of, and sharing of personal information that is collected by businesses. 

Allows consumers to know whether their personal data is sold or disclosed , to whom .

Allows opt-out right for sales of personal information

Right to deletion – to request a business to delete any personal information about a consumer collected from that consumer

Personal Data Protection Bill (PDP) – India 2018

This bill introduces various private and sensitive protection frameworks  like restriction on retention of personal data, Right to correction and erasure (such as right to be forgotten) , Prohibition and transparency of processing of personal data. It also classifies data fiduciaries  including certain social media intermediaries. 

The Bill amends the Information Technology Act, 2000 to delete the provisions related to compensation payable by companies for failure to protect personal data.

Other data privacy acts similar to GDPR 

  • South Korea’s Personal Information Protection Act  2011
  • Brazil’s Lei Geral de Proteçao de Dados (LGPD)  2020
  • Privacy Amendment (Notifiable Data Breaches) to Australia’s Privacy Act 2018
  • Japan’s Act on Protection of Personal Information 2017
  • Thailand Personal Data Protection Act (PDPA) 2020

Features offered by VOIP companies for Data privacy 

  • Access Control & Logging
  • Auto Data Redaction / Account Deletion policy 
  • SIEM (Security information and event management) alerts 
  • Information security , Encrypted Storage For Recordings & Transcripts
  • Disclosing all third party services that are involved in data processing too
  • Role Based Access Control and 2 Factor Authentication
  • Data Security Audits and appointing  data protection officer to oversee GDPR compliance

Against Robocalls and SPIT ( SPAM over Internet Telephony)

 2009 Truth in Caller ID Act 

Telephone Consumer Protection Act of 1991

Implementation of Do not call registry against use of robocalls, automatic dialers, and other methods of communication

Do-Not-Call Implementation Act of 2003

if a business has an established relationship with a customer, it can continue to call them for up to 18 months. If a consumer calls the company, say, to ask for information about the product or service, the company has three months to get back to him.

if the customer asks to not receive calls, the company must stop calling, or be subject to fines.

Exemptions – Calls from a not-for-profit B organisation , informational messages as flight cancellations , Calls from sales and debt collectors etc

Personal Data Privacy and Security Act 2009

Implemented to curb  identity theft and computer hacking. Sensitive personal identifiable information includes : victim’s name, social security number, home address, fingerprint/biometrics data, date of birth, and bank account numbers.

Any company that is breached must notify the affected individuals by mail, telephone, or email, and the message must include information on the company and how to get in touch with credit reporting agencies

If the breach involves government or national security , company must also contact the Secret Service within fourteen days 

TRACED Act (Telephone Robocall Abuse Criminal Enforcement and Deterrence) 2019

Canadian Radio-television and Telecommunications Commission (CRTC) 2018 -32

A solution mechanism has already been standardised and active in adoption called STIR / SHAKEN ( Secure Telephony Identity Revisited / Signature-based Handling of Asserted information using toKENs) described in another article here.

Emergency services 

FCC E911 E911 / VoIP E911 rules

Unlike traditional telephone connections, which are tied to a physical location, VOIP’s packet switched technology allows a particular number to be anywhere making it more difficult for it to reach localised services like emergency numbers of Public Safety Answering Points (PSAPs) . Thus FCC regulations as well as the New and Emerging Technologies 911 Improvement Act of 2008 (NET 911 Act), interconnected VoIP providers are required to provide 911 and E911 service. 

Ref : 

CLI/NCLI, Robocalls and STIR/SHAKEN

To understand the need for implementing an identification verification technique in Internet protocol based network to network communication system , we need to evaluate the existing problem plaguing the VoIP setup .

What is Call ID spoofing ? 

Vulnerability of existing interconnection phone system which is used by robo-callers to mask their identity or to make it appear the call is from a legitimate source, usually originates from voice-over-IP (VOIP) systems.

In this context understand the Caller Line identification CLI/ NCLI techniques used by VoIP and OTT( over the top) providers today.

CLI (Caller Line Identification)

If call goes out on a CLI route ( White Route ) the received party will likely see your callerID information

  • Lawful – Termination is legal on the remote end ie abiding country’s telco infrastructure and stable
  • Expensive – usually with direct or via leased line (TDM) interconnections with the tier-1 carriers.

Non-CLI (Non-Caller Line Identification)

The Caller ID is not visible at the call
If call goes out on a Non-CLI route (Grey Route) goes out on a non-CLI routes they will see either a blocked call or some generic number.

  • Unlawful – questionable legality or maybe violating some providers AUP(Acceptable Use Policy ) on the remote end.
  • Cheaper – low quality , usually via VoIP-GSM gateways

Example include robocalls , tele-marketting / spam etc which are unwilling to share their Caller Id for call receiver, to not be blocked or cancelled.

To overcome the problem of non-verifiable spam , robocalls a suite of protocols and procedures are proposed that can combat caller ID spoofing on VOIP and connected public telephone networks.

STIR/SHAKEN

Secure Telephony Identity Revisited / Signature-based Handling of Asserted information using toKENs

Used by robocallers to mask their identity or to make it appear the call is from a legitimate source
usually orignates from voice-over-IP (VOIP) systems

STIR

Standards developed by the Internet Engineering Task Force (IETF) 

For telecommunication service providers implement  certificate management system to create and manage the public and private keys, digital certificates used to sign and verify Caller ID details. 

Adds information to the SIP headers that allow the endpoints along the system to positively identify the origin of the data , such as JSON web tokens encrypted with the provider’s private key, encoded using Base64,

There are three levels of verification, or “attestation”

  • A : Full Attestation
    indicates that the provider recognizes the entire phone number as being registered with the originating subscriber.
  • B : Partial Attestation
    call originated with a known customer but the entire number cannot be verified,
  • C : Gateway Attestation
    call can only be verified as coming from a known gateway

How can the Public Key Infrastructure be used ? 

In an interconnection network , each telephone service provider will obtain its digital certificate from a certificate authority (CA)  that is trusted by other telephone service providers. Calling party signs the SIP Header  caller ID as legitimate . The called party verifies that the calling number is authentic

STIR

Originating service provider’s encrypted SIP Identity Header includes the following data:

  1. Attestation level
  2. Date and Time
  3. Calling and Called Numbers
  4. Orig ID for analytics and/or traceback purposes among others
  5. Location of certificate repository
  6. Signature
  7. Encryption algorithm

FCC has also assigned the role of a Secure Telephone Identity Policy Administrator (STI-PA) which oversees that CAs do not provide certificate to spoofing robocallers and enforce the framework for STIR /SHAKEN .

Sample Identity header in SIP requst

INVITE sip:bob@biloxi.example.org SIP/2.0
Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds8
To: Bob
From: Alice ;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Max-Forwards: 70
Date: Thu, 21 Feb 2002 13:02:03 GMT
Contact:
Identity:
"ZYNBbHC00VMZr2kZt6VmCvPonWJMGvQTBDqghoWeLxJfzB2a1pxAr3VgrB0SsSAaifsRdiOPoQZYOy2wrVghuhcsMbHWUSFxI6p6q5TOQXHMmz6uEo3svJsSH49thyGnFVcnyaZ++yRlBYYQTLqWzJ+KVhPKbfU/pryhVn9Yc6U="
Identity-Info: https://atlanta.example.com/atlanta.cer;alg=rsa-sha1
Content-Type: application/sdp
Content-Length: 147

v=0
o=UserA 2890844526 2890844526 IN IP4 pc33.atlanta.example.com
s=Session SDP
c=IN IP4 pc33.atlanta.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000

SHAKEN

STIR is based on the SIP protocol and is designed to work with calls being routed through a VOIP network. Since traditional endpoints like POTS and SS7 networks also should be covered under this call authenticity framework , SHAKEN was developed to manage call via IP-to-telephone gateways .

Developed by the Alliance of Telecommunications Industry Solutions (ATIS)

Working Steps  :

  1. When a call is initiated, a SIP INVITE is received by the originating service provider.
  2. Originating service provider verifies the call source and number to determine how to confirm validity.
    1. Full Attestation (A) — The service provider authenticates the calling party AND confirms they are authorized to use this number. An example would be a registered subscriber.
    2. Partial Attestation (B) — The service provider verifies the call origination but cannot confirm that the call source is authorized to use the calling number. An example would be a calling number from behind an enterprise PBX.
    3. Gateway Attestation (C) — The service provider authenticates the call’s origin but cannot verify the source. An example would be a call received from an international gateway.
  3. Create a SIP Identity header that contains information on the calling number, called number, attestation level, and call origination, along with the certificate thus caller ID “signed” as legitimate
  4. SIP INVITE with the SIP Identity header with the certificate is sent to the destination service provider.
  5. Destination service provider verifies the identity of the header and certificate.

Diagrammatic depiction of flow of how Telecom carriers to digitally validates authenticity before receiving or handoff through their network

SHAKEN

References

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