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786 bytes added, 15:56, 8 November 2020
Explain how SIP works
NAT is not officially supported, but generally can be made to work.
==How does SIP work?==
When your VoIP phone places a call, a message is sent on a control channel - this will conventionally be to the VoIP server's IP address on port 5060. When the call is answered, the two ends of the link have a conversation to agree on how to do the audio link (and video link too, if supported). The two ends of the link will agree on audio codecs to use (how the audio will be transmitted), and each end tells the other end where to send the audio packets to. The audio travels on a different channel from the control channel (using Real Time Protocol - RTP), and your VoIP phone will send something like "send your audio to this IP address, I'm listening for RTP on (for example) ports 5008 - 5012". Each end agrees, and you have a successful phone call.
==Why is NAT a problem with SIP?==

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