voip? 2

Discussion in 'Computing and Networks' started by Mathematics!, Jul 24, 2013.

  1. Mathematics!

    Thread Starter Senior Member

    Jul 21, 2008
    I am also wondering if I plug in a ip hardphone on the local network and assign it a static local ip address will I have to forward the port 5060 and/or 5061 (or something elses)

    Basically I can connect my hardphone directly to the sip account service provider by programming the information directly into it. But my problem then becomes how can people make calls back to me.

    I have dynamicdns setup and if that is not going to work because the ip hardphone only allows you to type in an ip address not a domain then I can get a static ip address either way I am assuming my problem will still be if I don't port forward to the router port that the ip hardphone is connected to then nobody will beable to reach my hardphone from the outside.

    Must I have an IP PBX or asterisks server running on the network or can I just connect it directly to the internet thru a router --- cable modem --- sip provider

    Just concerned about forwarding ports and static ip info for the uses of the ip hardphone

    I know the softphone applications allow me not to have to forward ports like google voice plugin provided that I am connected/signed in. Ofcourse in this case I am initial make the connection so it is still considered an outgoing connection not an initiated incoming connection that would require port forwarding. Though when it is a ip hardphone I am unsure about if it is an incoming or outgoing connection... I would assume if the ip hardphone was signed into a google voice/gmail account it wouldn't need to forward ports either duno.
    Anybody with some experience in this area
  2. Litch


    Jan 25, 2013
    No port forwards required.

    Your IP phone will register with your VoIP provider (Initiate an outgoing connection via UDP 5060). Your local gateway will then record that "conversation" and allow the VoIP provider's SIP packets back in to that IP phone.

    The SIP protocol has mechanisms to keep that "conversation" active so the router won't forget it, thus when call comes in from the VoIP provider, the INVITE message (The packet that starts a call) goes straight to your IP phone and presto.

    Just make sure your IP phone is set to send "SIP Keepalives" aka "OPTION messages" every 300 seconds (for example). If you find that incoming calls fail, try reduce that time value.