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Protocol

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Details

SIP (Session Initiation Protocol) is a multi-purpose protocol to initiate, maintain and terminate a communication
session between 2 partners in an IP network. It can be used for many communication forms and protocols, most common
usage is dialling for VoIP or VoLTE.

SIP can initiate arbitary media streams, like video for conferences or audio for Telephony, even multiple streams.
For VoIP it initiates 2 streams, a RTP stream for the audio data and RTCP for the stream control and status feedback.

SIP uses a header / body concept similar to HTTP and SMTP, it even adopts header variables of both protocols, but not all.
It can even have a text message payload in its body, in which case no further stream needs to be initiated at all.

The standard port for SIP is Port-5060-udp and Port-5060-tcp. A secure variant of SIP is SIPS

FRITZ!OS

In FRITZ!OS SIP is performed by voipd using the GNU libraries libosip2.so and libosipparser2.so with AVM's libsiplib.so.

For every Fritzbox with configured Telephony Port-5060-udp and Port-5060-tcp are open to the Internet.
This means it could be accessed without the need of a VoIP-Provider, unless it's no public IP (CGNAT) or your ISP blocks it.
SIP just needs the target IP, which could be acquired by DNS, ENUM, DynDNS or MyFRITZ!.

For client Fritzboxes in the LAN there's a DNS proxy on Port-7077-udp to use the DNS servers dedicated for VoIP. This is
neceessary since these servers may differ from the Internet DNS servers, see the DNS-Config section of the voipd article.

See also the RTP and RTCP articles.

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