Phone numbers and "SIP URIs like
email@example.com" are different kinds of URIs. A phone number can be represented as a URI like this:
whereas a "SIP URI like
firstname.lastname@example.org" can be represented as a URI like this:
Many SIP user agents don't actually use
tel: URIs. Instead they just take the number that the user dialed and stick into a SIP URI as the local-part and use the configured proxy server as the host-part, like this:
sip: URIs can be used directly to route a call over an IP network: just lookup the host-part of the SIP URI in DNS and forward the call to the location you find in the NAPTR/SRV/A/AAAA records.
tel: URIs must be routed according to rules configured in the SIP proxy. Examples of such rules could be:
- Send the call over a circuit-switched PSTN channel to the phone number given in the URI
- Forward the call to another SIP proxy, after possibly translating the phone number or even changing the URI from a
tel: URI to a
- Look up the phone number in DNS under
e164.arpa and translate the URI as per the NAPTR records found there.
All of this is unrelated to sending DTMF tones through a call. That can only happen after a call is established, and the SIP proxies, switches, and routers on the signaling path generally won't pay any attention to them (they're just media content, even if they're represented as RFC2833 events rather than as audio).