How does the sip protocol support the connection?

Question

SIP

There is an asterisk, ip phones are connected to it via nat. PBX constantly loses connection with one of the phones.

Contact:  3632/sip:[email protected]:32806            9b15fe183c Unavail         nan

And further:

Endpoint:  3632/3632                                            Unavailable   0 of 1
    OutAuth:  3632/3632
     InAuth:  3632/3632
        Aor:  3632                                               5
      Contact:  3632/sip:[email protected]:32806        9b15fe183c Unavail         nan

This is only with one phone. All others are working fine.

At the same time, outgoing calls from this phone work, when outgoing, it becomes available.

Tell me how to check the availability of phones from the asterisk? It's udp, so there are no sessions. The station cannot ping the phone, the phone is behind the nat. So the phone does not send any packets? Are these packets sent on port 5060 or RTP ports?

Answer 1

This is called registration, it goes through the signaling channel (5060 tcp / udp, as you configure)
If the registration flies, you need to try to reduce the interval.

Answer 2

To keep ports open on NAT, you can use:
- Decreased Expire Time on registration. For example, set 30s and the phone will be forced to re-register every 30 seconds.
- Sending "keep alive" packages. At the transport level in UDP, this is not possible, so the functionality works at the application level: the phone and the station exchange service packets, for example, OPTIONS. There is not much semantic load in them, their task is to maintain the session on NAT + show the station the availability of the phone.
Different NATs implement the SIP ALG mechanism differently, so there may be problems with it. If the problem phone is behind a separate NAT, this is worth paying attention to.