VoIP - Calling With a SIP URI: Difference between revisions

Latest revision as of 15:25, 23 July 2019

Calling a SIP URI with the 'Ekiga' softphone

With SIP, it's possible to make data calls over the Internet to SIP accounts by using URIs of the form sip:user@domain. The user field can be a name or a phone number. Such calls don't use the normal phone networks, and aren't charged as phone calls (although you may have to pay for the data used, like any other Internet traffic).

For example:

 * AAISP sales can be contacted at sip:sales@aa.net.uk
 * AAISP support can be contacted at sip:support@aa.net.uk
 * AAISP accounts an be contacted at sip:accounts@aa.net.uk
 * the acclaimed 'Lenny 'service is supposedly available at sip:13475147296@in.callcentric.com

Incoming calls can be made from the Internet to an AAISP number by using a SIP URI such as:

sip:number@aa.org.uk

A call to a URI like this will be delivered just like a normal incoming call - except the caller won't incur any call charges.

The format of the AAISP number should be in +44 format, e.g. to call AAISP support you'd use:

sip:+443333400999@aa.org.uk

You do need to use the hostname aa.org.uk as we use SRV records to direct the call, so using an IP address is not supported.
The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front of the number (which may confuse some caller ID display units), and not passed on if the call is diverted.
So, for example, without any configuration or account details a softphone may call your AAISP provided VoIP number, as shown in the image here.
As mentioned, this does require the client (the phone you're calling from) to support SRV records.

Clients

Most SIP clients which support calling SIP URIs are softphones. When testing, some clients can call SIP URIs without any configuration, other clients can call SIP URIs but require that a SIP account is defined (and perhaps enabled) although they don't actually use the account to place the call. There are also other clients which appear to support calling SIP URIs but which strip off the domain and call the number part as a chargeable phone call.

For testing clients, it can be useful to have a free SIP account which can't place external calls (e.g. linphone) - that way one can be sure that the client isn't making chargeable calls.

Known working

You may need to tell some of your contacts how to call you using a sip: URI. To help, the following SIP clients are known to work:

Softphones

Ekiga (Linux)
Linphone 4.0.1 (Android)
Twinkle (Linux) - although currently (Dec 2018) there's a problem with SRV records when using systemd-resolved (on Ubuntu and Debian at least). Need to do

systemctl stop systemd-resolved
systemctl disable systemd-resolved
systemctl mask systemd-resolved

add the line

dns=default

to the main section of /etc/NetworkManager/NetworkManager.conf and then

rm /etc/resolv.conf

to get NetworkManager to take over control of resolving.

Voipfone Softphone (Android) deserves a mention. It needs an account with voipfone.co.uk, but you can signup and try for free. Their softphone app works out of the box for calling SIP URIs - provided they don't have numeric usernames: so you can call AAISP's office (e.g. support@aa.net.uk) but you can't call sip:<number>@host (it'll call the number direct instead).

Known not-working

Linphone 3.6.1 (Linux) - doesn't use SRV DNS records, so can't contact @aa.org.uk SIP URIs.
CSipSimple, GS Wave - need a SIP account to be enabled, which they then use to place the call.
Zoiper (Android) - appears at first to work, but closer examination shows that it uses the SIP account to place calls (even when that account is disabled)

@@ Line 2: / Line 2: @@
 [[File:VOIP-sipURIcall.png|thumb|Calling a SIP URI with the 'Ekiga' softphone]]
+With SIP, it's possible to make data calls over the Internet to SIP accounts by using URIs of the form ''sip:user@domain''.  The ''user'' field can be a name or a phone number. Such calls don't use the normal phone networks, and aren't charged as phone calls (although you may have to pay for the data used, like any other Internet traffic).
-Incoming calls can be made from the internet to your number by using a SIP URI such as:
+For example:
+  * AAISP sales can be contacted at sip:sales@aa.net.uk
+  * AAISP support can be contacted at sip:support@aa.net.uk
+  * AAISP accounts an be contacted at sip:accounts@aa.net.uk
+  * the acclaimed 'Lenny 'service is supposedly available at sip:13475147296@in.callcentric.com
+Incoming calls can be made from the Internet to an AAISP number by using a SIP URI such as:
  sip:number@aa.org.uk
-A call to a URI like this will be delivered just like a normal incoming call.
+A call to a URI like this will be delivered just like a normal incoming call - except the caller won't incur any call charges.
-The format of your number should be in +44 format, e.g. to call AAISP support you'd use:
+The format of the AAISP number should be in +44 format, e.g. to call AAISP support you'd use:
  sip:+443333400999@aa.org.uk
 *You do need to use the hostname ''aa.org.uk'' as we use SRV records to direct the call, so using an IP address is not supported.
-*The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front, and not passed on if the call is diverted.
+*The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front of the number (which may confuse some caller ID display units), and not passed on if the call is diverted.
-*So, for example, without any configuration or account details a softphone may call your A&A provided VoIP number, as shown in the image here.
+*So, for example, without any configuration or account details a softphone may call your AAISP provided VoIP number, as shown in the image here.
 *As mentioned, this does require the client (the phone you're calling from) to support SRV records.
+==Clients==
+Most SIP clients which support calling SIP URIs are softphones.  When testing, some clients can call SIP URIs without any configuration, other clients can call SIP URIs but require that a SIP account is defined (and perhaps enabled) although they don't actually use the account to place the call.   There are also other clients which ''appear'' to support calling SIP URIs but which strip off the domain and call the number part as a chargeable phone call.
+For testing clients, it can be useful to have a free SIP account which can't place external calls (e.g. [https://www.linphone.org/freesip/home linphone]) - that way one can be sure that the client isn't making chargeable calls.
+==Known working==
+You may need to tell some of your contacts how to call you using a sip: URI.  To help, the following SIP clients are known to work:
+===Softphones===
+* Ekiga (Linux)
+* Linphone 4.0.1 (Android)
+* Twinkle (Linux) - although currently (Dec 2018) there's a problem with SRV records when using systemd-resolved (on Ubuntu and Debian at least).  Need to do
+ systemctl stop systemd-resolved
+ systemctl disable systemd-resolved
+ systemctl mask systemd-resolved
+add the line
+ dns=default
+to the ''main'' section of ''/etc/NetworkManager/NetworkManager.conf''
+and then
+ rm /etc/resolv.conf
+to get NetworkManager to take over control of resolving.
+* Voipfone Softphone (Android) deserves a mention.  It needs an account with voipfone.co.uk, but you can signup and try for free.  Their softphone app works out of the box for calling SIP URIs - provided they don't have numeric usernames: so you can call AAISP's office (e.g. support@aa.net.uk) but you can't call sip:<number>@host (it'll call the number direct instead).
+==Known not-working==
+* Linphone 3.6.1 (Linux) - doesn't use SRV DNS records, so can't contact @aa.org.uk SIP URIs.
+* CSipSimple, GS Wave - need a SIP account to be enabled, which they then use to place the call.
+* Zoiper (Android) - appears at first to work, but closer examination shows that it uses the SIP account to place calls (even when that account is disabled)
 [[Category:VoIP Features]]