SIP and TEL URLs
If the application requests a call to a telephone or a computer, RTC creates a SIP INVITE with the URL specified in the IRTCSession::AddParticipant method. The URL entered in this method can be SIP or TEL and can include an e-mail address, IP address, DNS name, or canonical telephone number. If a TEL URL is requested, RTC creates a SIP INVITE with the to: header set to 'to: phone-number'. RFC 2806 defines the formats for valid TEL URLs; however, RTC supports only a subset of those formats. The application is responsible for creating a valid TEL URL, and the number should follow the canonical phone number format. RTC strips off the qualifiers and sends only the canonical number to the gateway, which ultimately determines if the phone number is acceptable.
The following table shows examples of valid TEL URLs.
| TEL URL | Explanation |
|---|---|
| TEL: +1-425-555-0123 | TEL URL with global phone number format |
| TEL: +###-###-####### | TEL URL with global phone number format ('#' represents the digits in the actual phone number) |
The format for SIP URLs are given in RFC 2543 (SIP). The following table shows examples of valid SIP URLs.
| SIP URL | Explanation |
|---|---|
| SIP:someone@10.1.2.3 | SIP URL with IP address. |
| SIP: someone@example.com;transport=TCP | SIP URL with transport specified as TCP. The default transport is UDP if none is specified. |
| SIP: someone@example.com | Basic SIP URL. |
| SIP:+12225550189@example.com;user=phone | SIP URL with global phone number and specified gateway. |
| SIP: +1 206 555-0146@10.2.3.4;user=phone | SIP URL with global phone number format and specified gateway. |
Note that a SIP URI must contain legal characters as specified in RFC 3261.