Be more explicit about the local SIP port
VoIP using SIP has a control channel, which is used for registering with the VoIP server and for initiating outgoing calls and notification of incoming calls. The VoIP server normally listens on UDP port 5060. When a call is being setup and when the call is active, an additional audio channel is used.
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 being setup, 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.
An incoming call works the same way. It's important to note that the VoIP server will send control messages to your local SIP port,
Here's an (edited) SIP trace of an outbound call being setup from 02083xxxxxx to 07973xxxxxx. The phone on public IP address 81.187.xx.xx sends: