Which protocols describe VoIP connections

SIP - Session Initiation Protocol

Like H.323, SIP is responsible for setting up, operating and clearing voice and video connections. Both point-to-point and point-to-multipoint connections can be controlled with it.
Although H.323 was there first, interest in SIP has been very high since standardization by the IETF (Internet Engineering Task Force). Communication is relieved of the usual TK mechanisms and limited to the essentials. Due to its simplicity, SIP is easier to understand and the implementation effort is lower.

The strong reference to other Internet protocols is unmistakable in SIP. SIP is based heavily on HTTP (Hypertext Transfer Protocol). This means that SIP telephony can be easily integrated into browser environments, web services, applications and devices. Because all information is transmitted in clear text, it is recommended to use the encrypted version of SIPS.
The transmission of the data packets follows the logic of IP networks, which is why one speaks of Voice over IP when using SIP.

SIP protocols

Attendees
G.711 / G.729 / G.723 / ...
SIPSAP
SDP
RTP
TCPUDP
IP
Data link
Physical link

SIP is a text-based protocol that remote stations use to control their connections. Connection-oriented communication is implemented in a packet-switching network using SIP. It works on the 5th layer of the OSI layer model. This makes it independent of the underlying transport layers. SIP uses the transport protocols TCP and UDP for transmission. With RTP, the media streams are transmitted in real time. In parallel to RTP, RTCP is used to exchange important control information about the RTP media stream between client and server.

SDP - Session Description Protocol

SIP only describes the signaling. The SDP (Session Description Protocol), via which the communication partners exchange connection information, is embedded in the SIP protocol. With SDP, for example, IP address, media description, codec, ports and sending direction are exchanged. With the help of this information, the communication partners establish a direct connection to one another.

addressing

SIP is designed for the global localization of users across the Internet. The participants are addressed with URL and DNS.
Each SIP subscriber has an address that is similar to an email address (UserID @ Domain). The front part is either a username or a conventional phone number. The rear part addresses the SIP network.

SIP system architecture

SIP is based on a combined client / server architecture. There are parallels to e-mail and Jabber systems. With these services there are end systems (clients) that can only communicate with each other via auxiliary systems (proxy, gateway or server).
User agent, proxy server, redirect server and the registrar are defined in SIP. The user agent (UAC) is the client who initiates the calls. The User Agent Server (UAS) is the server or proxy that mediates the calls.

SIP communication

SIP is responsible for setting up, operating and clearing voice and video connections. For this purpose, SIP provides several dialogs to set up a session between two participants (user agent). SIP communication works according to the client-server principle.

NAT problem when establishing a connection with SIP

Establishing a connection with SIP requires that the end devices can be reached via a public IP address (IPv4). If one of the end devices is behind a router with NAT (Network Address Translation), this end device cannot be reached from outside via IP.
Simple Traversal of UDP through NAT, STUN for short, is a protocol with which two SIP end devices with an integrated STUN client can bypass the restrictions of NAT in many cases with the help of a STUN server.

ISDN and SIP in comparison

SIP makes the telephone number independent of the telephone connection. SIP accounts can be distributed to any location and device. An incoming call can ring at the workplace, mobile phone and private connection at the same time. In comparison, with ISDN and analog connections, the phone number is always bound.

How secure is SIP?

SIP is very complex when fully supported. However, most implementations are not complete. In order to work they are kept quite simple. Without any security precautions and protocols, VoIP connections with SIP can be rerouted, intercepted and evaluated.
Since VoIP with SIP is usually connected to the Internet, this infrastructure is very easy to attack. Internet access can be disrupted with little effort. For example, SIP servers can be flooded with a mass of connection attempts and blocked.

Overview: SIP

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