Prof. Anirudha Sahoo KReSIT, IIT Bombay Voice over IP Prof. Anirudha Sahoo KReSIT, IIT Bombay 19 September 2018 IIT Bombay

Voice Over IP (VOIP) Transmission of digitized voice in IP network Enables telephone conversation to be carried over IP network (in part or end-to-end) Provides a toll bypass path for telephone calls Enables Telephony providers to provide cheaper service 19 September 2018

VOIP System (A typical PSTN system) PSTN gateway PSTN gateway IP Network PSTN Network gatekeeper PBX PBX (A typical PSTN system) (A typical (H323 based) VOIP system) 19 September 2018

VOIP System (cont.) (A (SIP based) VOIP system) IP Network CPE router PSTN Gateway LAN LAN SIP proxy PSTN Soft phone IP phone IP phone (A (SIP based) VOIP system) 19 September 2018

VOIP Session Protocols Session Protocols are used to setup VOIP connections Primarily two competing protocols where endpoints are intelligent H.323 SIP MGCP is the other protocol with master-slave model where the end points are slaves. 19 September 2018

H.323 Defines interworking of Call signaling Call control Media stream protocols to build a packet-based multimedia communication system Also defines network components that are used to build up such a communication system 19 September 2018

Components of H323 network 19 September 2018

H.323 Terminals An endpoint on the network which provides for real-time, two-way communications with another H.323 terminal, Gateway or MCU using H.323 protocol Communication can consist of any multimedia data (e.g. speech, data or video). 19 September 2018

Multipoint Control Unit (MCU) Used for conference control Content mixing. Consists of a mandatory Multipoint Controller (MC) Responsible for call signaling, conference control An optional Multipoint Processor (MP) Responsible for switching/mixing of media streams. Can do real-time transcoding of audio/video streams Transcoding is expensive, so usually done in dedicated hardware (e.g. dsp) 19 September 2018

Gateway Provides two way communication between terminals belonging to networks with different protocol stack H323 terminal Gateway ISDN SIP terminal Speech terminal 19 September 2018

GateKeeper (GK) Provides address translation and controls access to the network resources for H.323 terminals, Gateways and MCUs. Endpoints register themselves at a GK All H.323 endpoints registered to a single GK build an H.323 zone H.323 zones are independent of physical network topology Each zone has only one GK An optional element in H.323. Without GK, there will not be any BW/admission control in the network and address resolution has to be done via other mechanism 19 September 2018

GK functionality Address Translation End point register their H323 aliases (address) and IP address A GK translates H.323 aliases into call signaling IP addresses Multiple GKs can communicate to build a multi-zone address translation service 19 September 2018

GK functionality (cont.) Admission control/bandwidth control Every call in the zone gets authorized by GK for admission Call control Direct call signaling/control and GK routed call signaling/control Direct signaling GK routed signaling 19 September 2018

H323 protocol overview 19 September 2018

RAS 19 September 2018

Q.931 and H.245 19 September 2018

Audio Codecs 19 September 2018

H.323 Call Setup Three phases to establish a call Phase A : GK Communication (Address resolution and admission) Phase B : Call Signaling (Setup, CallProc, Alert , Connect etc.) Phase C : Call Control (Capability Exchange, open/close media streams) (slow start mode) 19 September 2018

H323 call setup Basic setup without any GK endpoint2 endpoint1 setup CallProc Alert connect RTP media 19 September 2018

H323 call setup with GK GK endpoint2 endpoint1 ARQ ACF setup setup CallProc CallProc ARQ ACF Alert Alert connect connect 19 September 2018

H.323 Call SETUP (multi GK) 19 September 2018

SIP Overview Application Layer Signaling Protocol Used to establish, modify, and terminate multimedia sessions Part of Internet Multimedia Architecture Can use UDP, TCP, TLS etc. Based on HTTP (Web) Similar text-based structure Uses URIs (Uniform Resource Indicators) Applications include (but not limited to): Voice, video, gaming, instant messaging, presence, call control, etc. 19 September 2018

A Short History of SIP Internet Engineering Task Force (IETF) protocol Inventors: M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg Became “Proposed Standard” and RFC 2543 in March 1999 in MMUSIC WG. Separate SIP WG established in September 1999. Now new SIPPING (applications) and SIMPLE (presence and instant messaging) WGs using SIP. RFC2543bis-09 I-D became RFC 3261 in June 2002 Added four new authors: G. Camarillo, A. Johnston, J. Peterson, and R. Sparks. Entire spec rewritten for clarity, but some new features Mostly backwards compatible with RFC 2543 19 September 2018

SIP Requests and Responses SIP Responses use a numerical code and a “reason phrase” Classes: 1xx Informational 2xx Final 3xx Redirection 4xx Client Error 5xx Server Error 6xx Global Failure SIP Request types are called “methods” Methods in base spec: INVITE ACK OPTIONS CANCEL BYE REGISTER Example: 404 Not Found 19 September 2018

SIP Uniform Resource Indicators (URIs) Same form as email addresses: user@domain Two URI schemes: sip:henry@siptest.wcom.com is a SIP URI Most common form introduced in RFC 2543 sips:henry@siptest.wcom.com is a Secure SIP URI New scheme introduced in RFC 3261 Requires TLS over TCP as transport for security Two types of SIP URIs: Address of Record (AOR) (identifies a user) sip:henry@wcom.com (Needs DNS SRV records to locate SIP Servers for wcom.com domain) Fully Qualified Domain Name (FQDN) or Contact (identifies a device) sip:henry@127.24.45.4 or sip:henry@cube43.lab.wcom.com (Which needs no resolution for routing) 19 September 2018

Related Protocols SDP – Session Description Protocol Used to describe media session. Carried as a message body in SIP messages. Is a text-based protocol Uses RTP/AVP Profiles for common media types Defined by RFC 2327 RTP – Real-time Transport Protocol Used to transport media packets over IP RTP adds a bit-oriented header containing: name of media source timestamp codec type sequence number Defined by H. Schulzrinne et al, RFC 1889. Profiles defined by RFC 1890. RTCP for exchange of participant and quality reports. 19 September 2018

SIP “Trapezoid” 19 September 2018 DNS Server Location Server DNS SIP Inbound Proxy Server Outbound Proxy Server SIP SIP SIP Media (RTP) User Agent A User Agent B 19 September 2018

SIP Elements – User Agents Capable of sending and receiving SIP requests. UAC – User Agent Client UAS – User Agent Server End Devices SIP phone PC/laptop with SIP Client PDA mobile phone PSTN Gateways are a type of User Agent DNS Server Location Server DNS SIP Inbound Proxy Server Outbound Proxy Server SIP SIP SIP Media (RTP) User Agent A User Agent B 19 September 2018

SIP Elements – Proxy Servers DNS Server Location Server Forward or “proxy” requests on behalf of User Agents Consult databases: DNS Location Server Types: Stateless Transaction Stateful Call Stateful No media capabilities Ignore SDP. Normally bypassed once dialog established, but can Record-Route to stay in path. DNS SIP Inbound Proxy Server Outbound Proxy Server SIP SIP SIP Media (RTP) User Agent A User Agent B 19 September 2018

SIP Elements – Other Servers Location Server Database of locations of SIP User Agents Queried by Proxies in routing Updated by User Agents by Registration DNS Server SRV (Service) Records used to locate Inbound Proxy Servers DNS Server Location Server DNS SIP Inbound Proxy Server Outbound Proxy Server SIP SIP SIP Media (RTP) User Agent A User Agent B 19 September 2018

SIP Message Details INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 First line of a SIP message is Start Line which contains: the method or Request type: INVITE (session setup request). the Request-URI which indicates who the request is for sip:bob@biloxi.com Note: Request-URI can be either an AOR or Contact (FQDN) This Request-URI is a AOR the SIP version number SIP/2.0 19 September 2018

SIP Message Details Via headers show the path the request has taken INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 Via headers show the path the request has taken Additional Via headers are inserted by each proxy in the path The Via headers are used to route responses back the same way branch parameter is a transaction-ID 19 September 2018

SIP Message Details INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 Max-Forwards is a count decremented by each proxy that forwards the request. When count goes to zero, request is discarded and 483 Too Many Hops response is sent. Used for stateless loop detection. 19 September 2018

SIP Message Details INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 Dialog (formerly called call leg) information is in headers: To tag, From tag, and Call-ID (Note: Not URIs) To and From URIs usually contain AOR URIs. All requests and responses in this call will use this same Dialog information. Call-ID is unique identifier usually composed of pseudo-random string “@” hostname or IP Address 19 September 2018

SIP Message Details CSeq Command Sequence Number INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 CSeq Command Sequence Number Initialized at start of call (1 in this example) Incremented for each subsequent request within the same dialog Used to distinguish a retransmission from a new request Also contains the request type (method) - INVITE 19 September 2018

SIP Message Details INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 Contact header contains a SIP FQDN URI for direct communication between User Agents If Proxies do not Record-Route, they can be bypassed If Record-Route is present in 200 OK, then a Route header is present in all future requests in this dialog. Contact header is also present in 200 OK response 19 September 2018

SIP Message Details INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 Content-Type indicates the type of message body attachment (others could be text/plain, application/cpl+xml, etc.) Content-Length indicates the octet (byte) count of the message body. Message body is separated from SIP header fields by a blank line. 19 September 2018

SDP Message Body Details v=0 o=Tesla 289084526 28904529 IN IP4 lab.high-voltage.org s=- c=IN IP4 100.101.102.103 t=0 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Version number (ignored by SIP) Origin (only version used by SIP - 28904529) Subject (ignored by SIP) Connection Data (IP Address for media - 100.101.102.103) Time (ignored by SIP) Media (type - audio, port - 49170, RTP/AVP Profile - 0) Attribute (profile - 0, codec - PCMU, sampling rate – 8000 Hz) 19 September 2018

SIP Response Details SIP/2.0 200 OK Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=24019385 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:bob@200.201.202.203> Content-Type: application/sdp Content-Length: 176 v=0 o=Bob 2452772446 2452772446 IN IP4 200.201.202.203 s=SIP Call c=IN IP4 200.201.202.203 t=0 0 m=audio 56321 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Via, To, From, Call-ID, & CSeq are all copied from request. To now has a tag inserted by UAS Contact and Message Body contain UAS information. 19 September 2018

SIP Call Flow Scenarios Simple Call - Successful Call Attempt - Unsuccessful Presence Subscription Registration Presence Notification Call Setup – Successful 19 September 2018

Simple SIP call (Successful) biloxy.com proxy atlanta.com proxy Bob’s softphone Alice’s softphone INVITE INVITE 100 Trying INVITE 100 Trying 180 ringing 180 ringing 200 ok 180 ringing 200 ok 200 ok ACK RTP Media BYE 200 ok 19 September 2018

SIP Call Setup Attempt Scenario DNS Server Location Server 1. A “dials” SIP AOR URI sip:B@wcom.com. User Agent A sends INVITE to outbound Proxy Server. 2. Outbound Proxy sends 100 Trying response. Inbound Proxy Server Outbound Proxy Server 1. INVITE Contact: A SDP A 2. 100 Trying User Agent A User Agent B (Not Signed In) 19 September 2018

SIP Call Setup Attempt Scenario DNS Server Location Server 3. DNS Query: wcom.com? 4. Response: 1.2.3.4 3. Outbound Proxy does DNS query to find proxy server for wcom.com domain 4. DNS responds with IP address of wcom.com Proxy Server Inbound Proxy Server Outbound Proxy Server 1. INVITE Contact: A SDP A 2. 100 Trying User Agent A User Agent B (Not Signed In) 19 September 2018

SIP Call Setup Attempt Scenario DNS Server Location Server 3. DNS Query: wcom.com? 4. Response: 1.2.3.4 5. Outbound Proxy sends INVITE to Inbound Proxy Server. 6. Inbound Proxy sends 100 Trying response. 5. INVITE Contact: A SDP A Inbound Proxy Server Outbound Proxy Server 6. 100 Trying 1. INVITE Contact: A SDP A 2. 100 Trying User Agent A User Agent B (Not Signed In) 19 September 2018

SIP Call Setup Attempt Scenario DNS Server Location Server 3. DNS Query: wcom.com? 4. Response: 1.2.3.4 7. LS Query: B? 8. Response: Not Signed In 7. Inbound Proxy consults Location Server. 8. Location Server responds with “Not Signed In.” 5. INVITE Contact: A SDP A Inbound Proxy Server Outbound Proxy Server 6. 100 Trying 1. INVITE Contact: A SDP A 2. 100 Trying User Agent A User Agent B (Not Signed In) 19 September 2018

SIP Call Setup Attempt Scenario DNS Server Location Server 3. DNS Query: wcom.com? 4. Response: 1.2.3.4 7. LS Query: B? 8. Response: Not Signed In 9. Inbound Proxy sends 480 Temporarily Unavailable response. 10. Outbound Proxy sends ACK response. 5. INVITE Contact: A SDP A Inbound Proxy Server Outbound Proxy Server 6. 100 Trying 1. INVITE Contact: A SDP A 9. 480 Temporarily Unavailable 10. ACK 2. 100 Trying User Agent A User Agent B (Not Signed In) 19 September 2018

SIP Call Setup Attempt Scenario DNS Server Location Server 3. DNS Query: wcom.com? 4. Response: 1.2.3.4 7. LS Query: B? 8. Response: Not Signed In 11. Outbound Proxy forwards 480 response to A. 12. A sends ACK response. 5. INVITE Contact: A SDP A Inbound Proxy Server Outbound Proxy Server 6. 100 Trying 1. INVITE Contact: A SDP A 9. 480 Temporarily Unavailable 10. ACK 2. 100 Trying 11. 480 Temporarily Unavailable 12. ACK User Agent A User Agent B (Not Signed In) 19 September 2018

User Agent B (Not Signed In) SIP Presence Example DNS Server Presence Server 1. A wants to be informed when B signs on, so sends a SUBSCRIBE 2. Outbound Proxy forwards to Inbound Proxy 3. Inbound Proxy forwards to B’s Presence Server 3. SUBSCRIBE 2. SUBSCRIBE Inbound Proxy Server Outbound Proxy Server 1. SUBSCRIBE User Agent A User Agent B (Not Signed In) 19 September 2018

User Agent B (Not Signed In) SIP Presence Example DNS Server Presence Server 3. SUBSCRIBE 4. 200 OK 4. Presence Server authorizes subscription by sending a 200 OK. 5. & 6. 200 OK proxied back to A. 2. SUBSCRIBE Inbound Proxy Server Outbound Proxy Server 5. 200 OK 1. SUBSCRIBE 6. 200 OK User Agent A User Agent B (Not Signed In) 19 September 2018

User Agent B (Not Signed In) SIP Presence Example DNS Server Presence Server 7. Presence Server sends NOTIFY containing current presence status of B (Not Signed In). 8. and 9. NOTIFY is proxied back to A. 10. A acknowledges receipt of notification with 200 OK. 11. & 12. 200 OK is proxied back to B’s Presence Server. 7. NOTIFY <Not Signed In> 12. 200 OK 8. NOTIFY <Not Signed In> Inbound Proxy Server Outbound Proxy Server 11. 200 OK 9. NOTIFY <Not Signed In> 10. 200 OK User Agent A User Agent B (Not Signed In) 19 September 2018

SIP Registration Example DNS Server Location Server 2. Update database: B = B@2.3.4.5 1. B signs on to his SIP Phone which sends a REGISTER message containing the FQDN URI of B’s User Agent. 2. Database update is sent to the Location Server Outbound Proxy Server Outbound Proxy Server 1. REGISTER Contact: B@2.3.4.5 User Agent A User Agent B 19 September 2018

SIP Registration Example DNS Server Location Server 3. Location Server database update is confirmed. 4. Registration is confirmed with a 200 OK response. 2. Update database: B = B@2.3.4.5 3. OK Outbound Proxy Server Outbound Proxy Server 1. REGISTER Contact: B@2.3.4.5 4. 200 OK Contact: B@2.3.4.5 User Agent A User Agent B 19 September 2018

SIP Presence Example DNS Server Presence Server 13. Presence Server learns of B’s new status from the Location Server and sends a NOTIFY containing new status of B (Signed In). 14. & 15. NOTIFY is proxied back to A. 16. A acknowledges receipt of notification with 200 OK. 17. & 18. 200 OK is proxied back to Presence Server. 13. NOTIFY <Signed In> 18. 200 OK 14. NOTIFY <Signed In> Inbound Proxy Server Outbound Proxy Server 17. 200 OK 15. NOTIFY <Signed In> 16. 200 OK User Agent A User Agent B 19 September 2018

SIP Call Setup Attempt Scenario DNS Server Location Server 1. to 5. A retries INVITE to B which routes through two Proxy Servers. 6. Location Server responds with the FQDN SIP URI of B’s SIP Phone. 7. Inbound Proxy Server forwards INVITE to B’s SIP Phone. 5. LS Query: B 6. Response: sip:B@2.3.4.5 3. INVITE Contact: A SDP A Inbound Proxy Server Outbound Proxy Server 4. 100 Trying 1. INVITE Contact: A SDP A 2. 100 Trying 7. INVITE Contact: A SDP A User Agent A User Agent B 19 September 2018

SIP Call Setup Scenario DNS Server Location Server 8. User Agent B alerts B and sends 180 Ringing response. 9. & 10. 180 Ringing is proxied back to A. 9. 180 Ringing Inbound Proxy Server Outbound Proxy Server 10. 180 Ringing 8. 180 Ringing User Agent A User Agent B 19 September 2018

SIP Call Setup Scenario DNS Server Location Server 11. B accepts call and User Agent B sends 200 OK response. 12. & 13. 200 OK is proxied back to A. 9. 180 Ringing Inbound Proxy Server Outbound Proxy Server 12. 200 OK Contact: B SDP B 10. 180 Ringing 13. 200 OK Contact: B SDP B 11. 200 OK Contact: B SDP B 8. 180 Ringing User Agent A User Agent B 19 September 2018

SIP Call Setup Scenario DNS Server Location Server 14. ACK is sent by A to confirm setup call bypassing proxies. Media session begins between A and B! 9. 180 Ringing Inbound Proxy Server Outbound Proxy Server 12. 200 OK Contact: B SDP B 10. 180 Ringing 13. 200 OK Contact: B SDP B 11. 200 OK Contact: B SDP B 8. 180 Ringing 14. ACK Media (RTP) User Agent A User Agent B 19 September 2018

Other SIP Methods in RFC 3261 CANCEL Used to terminate a pending session INVITE sent, and 1xx received, but no “final response” Can be sent by a proxy or User Agent Useful for “forking proxy” Parallel search using multiple registration Contacts. First successful wins, rest are cancelled. OPTIONS Used to query capabilities of a Server or User Agent 19 September 2018

References International Telecommunications Union. ITU-T Recommendation H.323 J. Rosenberg et al., “SIP: Session Initiation Protocol” – Request for Comments 3261 B. Goode, “Voice over Internet Protocol (VOIP)” – Proceedings of the IEEE, vol. 90, no. 9, September 2002. http://www.packetizer.com http://www.pulver.com/ http://www.voip-info.org http://voip.internet2.edu 19 September 2018

Introduction to VOIP Security 19 September 2018

VOIP Security As in Data, VOIP Security is very important in today’s context People would expect similar level of security as PSTN system Physical access to PSTN/compromising PBX is required to breach security in PSTN But breaching the securiy in IP based call is much easier. Hence the VOIP protocols define security framework 19 September 2018

Security in H323 Call Establishment (H.225) Security Secured mode of communication should be used (e.g.IPSEC or TLS) before exchange of call connection message Call Control (H.245) Security H245 channel is secured using negotiated privacy mechanism Typically done during H225 signaling Media Stream Privacy Media stream encoded using algorithm and keys as presented during H245 signaling 19 September 2018

Security in H323 H.235 specifies security aspects of H323 Various security profiles recommended by H.235 19 September 2018

Annex D Baseline Security Uses symmetric cryptography technique Applicable where subscribed passwords/symmetric keys can be assigned to H323 entities. Passwords used for authentication Symmetric keys used for encryption Confidentiality of media stream provided by symmetric encryption 19 September 2018

Annex D Baseline Profile RAS H225 H245 Authentication Shared secret (password) HMAC-SHA1-96 Integrity Key Management Subscription based password assignment 19 September 2018

Annex D Baseline Profile Uses CryptoH323Token field to send encrypted message HMAC-SHA1-96 algorithm is used on the entire message which includes a sequence number (monotonically increasing), timestamp The GK upon receiving the encrypted message verifies the authenticator based on Liveness of the timestamp Matching of the authenticator in the message with that computed by the GK *HMAC – Hash Message Authentication Code 19 September 2018

Annex E Signature Profile Uses asymmetric crypto techniques Certificates and Digital signatures used to provide authentication and integrity 19 September 2018

Annex E Signature Profile RAS H225 H245 Authentication SHA1/MD5 Digital signature Integrity Key Management Certificate allocation 19 September 2018

Secured RAS Example H323 terminal gatekeeper Digital signature = MD5-RSA( monotinally increasing seq num, timestamp, sender id, receiver id, and few other fields) RRQ, digital signature Gatekeeper verifies signature RCF 19 September 2018

Annex D Voice Encryption Profile Provides confidentiality of the actual voice media stream being transmitted. Uses Diffie-Hellman Key exchange during connection set up This key is used to protect media keys that are used to encrypt media (RTP) packets. Encryption algorithm chosen are 56bit DES, 56-RC2, 168 bit triple DES, AES 19 September 2018

Use of IPSec/TLS in H323 Can be used to provide authentication and confidentiality at the IP layer This is independent of protocol run in the upper layer Only policies at the end points should be configured to use IPSec/TLS. RAS, H225 and H245 signaling can be done over IPSec/TLS. 19 September 2018

Security in SIP SIP specification (RFC 3261) specifies security for SIP protocol. Distributed nature of the protocol makes it very susceptible to security compromise Hence the need to provide framework for secured SIP signaling and media 19 September 2018

Attacks and Threat Models Registration hijacking Attacker successfully impersonates a party who can register and de-register De-registers all the contacts for a URI and register himself as the contact Now the attackers gets all the SIP requests Impersonating server A compromised server can easily be used to direct request or consume resources illegally 19 September 2018

Attacks and Threat Models Tampering with message bodies Malicious proxy can change the message bodies Can redirect RTP packets to a wiretapping device Can launch “man-in-the-middle” attack to compromise the integrity of the message Tearing down sessions If an attacker gets holds of the essential parameters of a session (To tag, From tag etc.), then it could forge a BYE request to end the session 19 September 2018

Attacks and Threat Models Denial of Service Attacker can create bogus request that contain falsified source address and a corresponding via header set to a target host If these requests are sent to a large number of SIP elements, then the host would be inundated with DOS traffic Attacker can de-register legitimate users from a proxy which would prevent any new sessions for the users Attacker can register lots of bogus users to deplete memory and disk resources 19 September 2018

HTTP Authentication SIP uses Digest authentication of http Challenge-response paradigm Provides message authentication and replay protection, but not message integrity or confidentiality Each authentication is meaningful for a particular domain (or realm) Any time a proxy or UA receives a message, it may challenge the initiator of the request to provide assurance of its identity Basic method of http (user name passwd passed in clear text) is not allowed in SIP UAS uses 401 (not authorized) and proxy uses 407 (proxy authentication required) to challenge the initiator 19 September 2018

www-authenticate: Digest HTTP Digest UAC UAS INVITE Does not have credential www-authenticate: Digest nonce : o 401 (not authorized) INVITE (contains Digest) Checks the credential 100 trying 19 September 2018

HTTP Digest UAS should not challenge two requests ACK CANCEL Does not have a response UAC sending ACK would copy all the credentials sent in INVITE CANCEL Cannot be resubmitted Generally accepted by a server if it comes from the same hop that sent the request being cancelled Requires transport or network layer security support 19 September 2018

S/MIME SIP message carries MIME bodies MIME standard includes mechanism for securing MIME contents (integrity and confidentiality) S/MIME certificate used to identify end user This certificate also has key that is used to encrypt the bodies of the message. Bodies signed with private key of the sender and encrypted with public key of the recipient. 19 September 2018

Use of Network and Transport Layer Security Transport and network layer security encrypts SIP signaling traffic ensuring message confidentiality and integrity. Two popular options are TLS and IPSec IPSec does not require tight integration with SIP Used in architectures in which a set of hosts have trust relationship TLS provides security at the transport layer Used on hop by hop basis : Each hop can extend TLS session to the next hop to achieve end-to-end security 19 September 2018

SIPS URI Scheme sips: can be used instead of sip: as AOR of a user When used, it signifies that each hop over which the request is forwarded must be secured with TLS (until the domain portion of the request URI) Once it reaches the domain in question, it is handled by the local security and routing policy 19 September 2018

Firewall/NAT Issues Firewall provide a central location for deploying security policies. Relieves end points of maintaining security policies. Takes a lot of burden out of the VoIP infrastructure NAT is a powerful technology that can be used to hide internal network addresses and enable several endpoints in the network to share the same external address From security view point, makes internal IP addresses less accessible Attack against the network need be focused only at NAT But these come at a Price Few issues crop up for VoIP in NAT/Firewall environment 19 September 2018

VoIP with NAT/Firewall 19 September 2018 Source: National Institute of Standards and Technology

Firewall/NAT Issues Firewall issues NAT issues RTP ports used by VOIP are dynamically decided during signaling Hence firewalls cannot statically open the ports Opening all the RTP ports is not a wise thing NAT issues The IP address and port information is carried in the payload Once H323/SIP messages traverse a NAT, the IP address would not be valid (NAT translates the IP address only in the IP header) 19 September 2018

Firewall/NAT Solutions Stateful Firewalls Can investigate application data in a packet Open the dynamic RTP ports (pinhole) by peeking into the H.323/SIP signaling messages Proxy placed at the border between two domains Proxy would terminate sessions with both the hosts and relay signaling as well as RTP media streams 19 September 2018

Firewall/NAT Solutions Application Level Gateways (ALG) Examines and modifies application payload contents to allow traversal through NAT/Firewall Simple Traversal of UDP through NAT (STUN) A simple client server protocol STUN client (usually embedded in applications) sends binding request STUN client sends response which carries public IP and port mapping generated by NAT Application needs to listen at the IP addr and port from which binding request was sent Packets from remote hosts destined to public IP and port will be received by the application 19 September 2018

References International Telecommunications Union. ITU-T Recommendation H.235 J. Rosenberg et al., “SIP: Session Initiation Protocol” – Request for Comments 3261 National Institute of Standards and Technology, “Security Considerations for Voice over IP Systems”- Special publication 800-58. 19 September 2018

Slides available at http://www.it.iitb.ac.in/~sahoo/voip_security.ppt 19 September 2018