WLAN and IP > VOIP > VOIP SIP Phone Configure

VOIP SIP Phone Configure

VOIP/SIP phones can be physical phones connected directly to a router (Right: Siemens C450IP) or software run on a computer (Left: XLite softphone)

VOIP SIP Phone Set Up

To configure a SIP phone you need to have an account with a SIP provider. A SIP provider keeps your SIP phone number stored in it's registry and sets up the calls to and from your SIP phone.

Some popular free SIP providers are: www.sipphone.com and www.freeworlddialup.com (FWD).

When you sign up for a free account, you will receive a SIP phone number, for sipphone.com it will be in a form of 1-747-123-4567 (altho it looks like a landline number, it is not). FWD has a 6 digit number like 987654. (Note! It can be possible a SIP phone provider can issue a SIP number that contains letters (ex: voipbuster.com). Try to avoid this, as some hard SIP phones can only dial numbers!) A password of your choice is also required when subscribing.

You should also receive "Proxy Server" and "Registry Server" URL and port information. When a SIP call is placed it will be sent to the server in the form of:

<SIP Number>@<Register URL>:<Port>

ex: 17471234567@proxy01.sipphone.com:5060

The Domain, Proxy Server and Register Server are usually the same URL (altho they can be different URL's), and the default SIP port is usually 5060 (altho can be any port determined by SIP provider).

Necessary basic info for any SIP phone set up:

SIP Number: Given by SIP provider (ex: 17471234567)
Password: Your password when registering to SIP provider

Domain: Given by SIP provider (ex: proxy01.sipphone.com)

Proxy Server: Given by SIP provider (ex: proxy01.sipphone.com)
Proxy Server Port: Given by SIP provider (usually 5060 is default)

Registrar Server: Given by SIP provider (ex: proxy01.sipphone.com)
Registrar Port: Given by SIP provider (usually 5060 is default)
Reg Refresh/Period: How often in seconds your phone should send location info (default 180, altho the SIP server SBC may override this to a value that doesn't overload it's server)

Codec: G.711u/G.711a are the most common codecs, check that your SIP phone and the one you are calling both have it.

Optional: RTP Port: normally a random port, but a fixed port of 5004, 5006, 8000, 30000 or 60000 is also common
Optional: RTP packet: 20ms default for G.711 codec, G.722 is 30ms

IP Address: It may also help to give your SIP phone a fixed local IP address (ex: 192.168.1.30) since DHCP may cause problems. A fixed IP address is also useful if you later need to port open/forward from your router to the SIP phone.

STUN (Optional, Reccommended)
When calling outside a local network, the use of a STUN server may help to bypass NAT problems (ie: your SIP phone has a local 192.x.x.x IP address but the router has an external public IP address). You can use any STUN server regardless of who your SIP provider is (ex. you may use stun.fwdnet.com with a sipphone.com account). The default STUN port is 3478. Some free STUN servers are:

STUN Port: 3478

stun01.sipphone.com
stun.xten.com
stun.sipdiscount.com
stun.voipstunt.com
stun.voipbuster.com
stunserver.org
stun.ekiga.net

Refresh time in seconds is usually 60 (use 20 if experiencing NAT problems).

STUN can solve most NAT problems (~70%) but not all. If one STUN server fails to work, try another. If none work, try using an Outbound Server. One way or no audio is often a NAT problem. SIP signaling and RTP voice traffic are handled seperately, so it's possible the signalling is successful (ie: phone rings) but the RTP stream is unsuccessful (ie: no audio)
(Note: Most SIP phones support STUN, but not all)

OUTBOUND (or TURN) SERVER (Optional, Only if Necessary)
If the STUN server doesn't help, you can use a Outbound Server, which breaks the call into 2 legs: you call the Outbound server, and the outbound server calls the end number. If you are behind a Symmetric NAT, use of an outbound server may be necessary. If using an Outbound Server, disable STUN. One free outbound server is:

Outbound Proxy: fwdnat2.pulver.com:5082
Listen RTP Port: 8000
Listen SIP Port: 5082

 or

Outbound Proxy: fwdnat.pulver.com:5060
Listen RTP Port: 8000
Listen SIP Port: 5060

ICE is a method of automatically first attempting to use a STUN server, and in the case it fails, then to automatically use a TURN server.

(Note: Most SIP phones support the Outbound Server feature, but not all)

PORT OPENING / FORWARDING (Optional)
You may also use port opening and port forwarding in your router to open any blocked connections. SIP normally uses UDP. If you have several SIP phones each should have a different port assigned. RTP often uses dynamic ports and, depending on your phone, it may or may not be possible to fix a port for RTP.

TEST CALL
Once a SIP phone is properly configured, it should take less than a minute for your phone to be recognized as "Registered" by the SIP providers registrar (this should be shown on your phone as well)

To make a test call using a SIP phone configured with an outbound sipphone.com account, dial: ** to hear your phone number repeated to you. Or for a full list of test numbers: http://www.sipphone.com/numbers/

To make a test call using FWD dial 613 for an echo test, or 55555 for a live person.

If the test call is successful, you should be able to directly call other SIP numbers with the same SIP provider you have just enabled.

CALL EXTERNAL SIP NETWORKS
To call out from one SIP provider to another is possible (if it is supported by your SIP provider) by adding a prefix to the number. Your SIP provider should have a list of prefixes listed for SIP providers they have interoperability agreements with. This is called "peering". For example to call a FWD number from a sipphone.com phone, you must first enter a prefix of *393 and then the number

ex: to call the FWD number 987654 from another FWD account, it is simply dialed directly, but a sipphone.com user would have to dial *393987654

A comprehensive list of SIP provider prefixes (as well as SIP providers) can be found at: http://www.sipbroker.com/sipbroker/action/providerWhitePages

Note: the prefixes listed on SipBroker may be different than your SIP provider. You should check with your SIP provider for a list of external prefixes.

Alternatively, some SIP phones allow you to dial a number on an external SIP providers directly by calling in the full format <SIP Number>@<Register URL>:<Port>. For example, to call a FWD number 987654 from a sipphone.com account, it can be dialed: 987654@fwd.pulver.com:5060

Some SIP phones allow several SIP provider accounts to be registered on the same phone simultaneously. This only applies for incoming calls. For outgoing calls, only one SIP provider at a time can be selected.

Some SIP providers also allow calling PSTN (ie: normal fixed-line phones) numbers, but usually charge a  fee. VoipBuster allows free calls to certain countries PSTN's on a trial basis: http://www.voipbuster.com/en/free.html

TROUBLESHOOTING

There can be many reasons for failed SIP connections. NAT problems and improperly configured equipment are 2 common reasons. To further read about NAT transversal, read this whitepaper by Newport Networks: NAT Traversal for Multimedia Over IP

To help troubleshoot, a packet sniffer such as WireShark can help discover problems. Download WIreShark for free at: http://www.wireshark.org/.