Overview Of Voice over IP (VoIP) in SonicOS Enhanced
03/26/2020 198 13795
DESCRIPTION: Overview Of Voice over IP (VoIP) in SonicOS Enhanced.
RESOLUTION: Feature Description:
This article provides an overview of VoIP. It contains the following sections:
“What is VoIP?”
Firewall Requirements for VoIP
What is VoIP?
Voice over IP (VoIP) is an umbrella term for a set of technologies that allow voice traffic to be carried over Internet Protocol (IP) networks. VoIP transfers the voice streams of audio calls into data packets as opposed to traditional, analog circuit-switched voice communications used by the public switched telephone network (PSTN).
VoIP is the major driving force behind the convergence of networking and telecommunications by combining voice telephony and data into a single integrated IP network system. VoIP is all about saving cost for companies through eliminating costly redundant infrastructures and telecommunication usage charges while also delivering enhanced management features and calling services features.
Companies implementing VoIP technologies in an effort to cut communication costs and extend corporate voice services to a distributed workforce face security risks associated with the convergence of voice and data networks. VoIP security and network integrity are an essential part of any VoIP deployment.
The same security threats that plague data networks today are inherited by VoIP but the addition of VoIP as an application on the network makes those threats even more dangerous. By adding VoIP components to your network, you’re also adding new security requirements.
VoIP encompasses a number of complex standards that leave the door open for bugs and vulnerabilities within the software implementation. The same types of bugs and vulnerabilities that hamper every operating system and application available today also apply to VoIP equipment. Many of today's VoIP call servers and gateway devices are built on vulnerable Windows and Linux operating systems.
Firewall Requirements for VoIP:
VoIP is more complicated than standard TCP/UDP-based applications. Because of the complexities of VoIP signaling and protocols, as well as inconsistencies that are introduced when a firewall modifies source address and source port information with Network Address Translation (NAT), it is difficult for VoIP to effectively traverse a standard firewall. Here are a few of the reasons why.
• VoIP operates using two separate protocols - A signalling protocol (between the client and VoIP Server) and a media protocol (between the clients). Port/IP address pairs used by the media protocols (RTP/RTCP) for each session are negotiated dynamically by the signalling protocols. Firewalls need to dynamically track and maintain this information, securely opening selected ports for the sessions and closing them at the appropriate time.
• Multiple media ports are dynamically negotiated through the signalling session - negotiations of the media ports are contained in the payload of the signalling protocols (IP address and port information). Firewalls need to perform deep packet inspection on each packet to acquire the information and dynamically maintain the sessions, thus demanding extra firewall processing.
• Source and destination IP addresses are embedded within the VoIP signalling packets - A firewall supporting NAT translates IP addresses and ports at the IP header level for packets. Fully symmetric NAT firewalls adjust their NAT bindings frequently, and may arbitrarily close the pinholes that allow inbound packets to pass into the network they protect, eliminating the service provider's ability to send inbound calls to the customer. To effectively support VoIP it is necessary for a NAT firewall to perform deep packet inspection and transformation of embedded IP addresses and port information as the packets traverse the firewall.
• Firewalls need to process the signalling protocol suites consisting of different message formats used by different VoIP systems - Just because two vendors use the same protocol suite does not necessarily mean they will interoperate. To overcome many of the hurdles introduced by the complexities of VoIP and NAT, vendors are offering Session Border Controllers (SBCs). An SBC sits on the Internet side of a firewall and attempts to control the border of a VoIP network by terminating and re-originating all VoIP media and signalling traffic. In essence, SBCs act as a proxy for VoIP traffic for non-VoIP enabled firewalls. SonicWall security appliances are VoIP enabled firewalls that eliminate the need for an SBC on your network.
VoIP technologies are built on two primary protocols, H.323 and SIP.
H.323 is a standard developed by the International Telecommunications Union (ITU). It is a comprehensive suite of protocols for voice, video, and data communications between computers, terminals, network devices, and network services. H.323 is designed to enable users to make point-to-point multimedia phone calls over connectionless packet-switching networks such as private IP networks and the Internet.
H.323 is widely supported by manufacturers of video conferencing equipment, VoIP equipment and Internet telephony software and devices. H.323 uses a combination of TCP and UDP for signalling and ASN.1 for message encoding. H.323v1 was released in 1996 and H.323v5 was released in 2003. As the older standard, H.323 was embraced by many early VoIP players. An H.323 network consists of four different types of entities:
• Terminals- Client end points for multimedia communications. An example would be an H.323 enabled Internet phone or PC.
• Gatekeepers - Performs services for call setup and tear down, and registering H.323 terminals for communications. Includes: – Address translation. – Registration, admission control, and status (RAS). – Internet Locator Service (ILS) also falls into this category (although it is not part of H.323). ILS uses LDAP (Lightweight Directory Access Protocol) rather than H.323 messages.
• Multipoint control units (MCUs)- Conference control and data distribution for multipoint communications between terminals.
• Gateways- Interoperation between H.323 networks and other communications services, such as the circuit-switched Packet Switched Telephone Network (PSTN).
Session Initiation Protocol (SIP):
The Session Initiation Protocol (SIP) standard was developed by the Internet Engineering Task Force (IETF). RFC 2543 was released in March 1999. RFC 3261 was released in June 2002. SIP is a signaling protocol for initiating, managing and terminating sessions. SIP supports ‘presence’ and mobility and can run over User Datagram Protocol (UDP) and Transmission Control Protocol (TCP). Using SIP, a VoIP client can initiate and terminate call sessions, invite members into a conferencing session, and perform other telephony tasks. SIP also enables Private Branch Exchanges (PBXs), VoIP gateways, and other communications devices to communicate in standardized collaboration. SIP was also designed to avoid the heavy overhead of H.323. A SIP network is composed of the following logical entities:
• User Agent (UA) - Initiates, receives and terminates calls.
• Proxy Server - Acts on behalf of UA in forwarding or responding to requests. A Proxy Server can fork requests to multiple servers. A back-to-back user agent (B2BUA) is a type of Proxy Server that treats each leg of a call passing through it as two distinct SIP call sessions: one between it and the calling phone and the other between it and the called phone. Other Proxy Servers treat all legs of the same call as a single SIP call session.
• Redirect Server- Responds to request but does not forward requests.
• Registration Server- Handles UA authentication and registration.