SIP and TEL URIs
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8/28/2008
If an application that uses the RTC Client API initiates a call to a telephone or computer, the RTC Client API implementation creates a SIP INVITE message with the URI specified in the IRTCSession::AddParticipant method.
The SIP or TEL URI can include the following:
- E-mail address
- IP address
- DNS name
- Canonical telephone number
If a TEL URI is requested, the RTC Client API implementation creates a SIP INVITE with the to header set to the following: to: telephone-number.
The application is responsible for creating a valid TEL URI and the number should follow the canonical phone number format.
The RTC Client API implementation strips off the qualifiers and sends only the canonical number to the gateway. The gateway determines if the number is acceptable.
The following are examples of valid TEL URIs.
| TEL URL | Explanation |
|---|---|
TEL: +1-425-555-4567 |
TEL URI with global phone number format. |
TEL: +358-555-1234567 |
TEL URI with global phone number format. |
The format for SIP URIs is given in RFC 3261. The following are examples of valid SIP URIs.
| SIP URL | Explanation |
|---|---|
SIP:someone@10.1.2.3 |
SIP URI with IP address. |
SIP: someone@example.com;transport=TCP |
SIP URI with transport specified as TCP. The default transport is UDP if none is specified. |
SIP: someone@example.com |
Basic SIP URI. |
SIP: +1 206 555-1234@1.2.3.4;user=phone |
SIP URI with global phone number format and specified gateway. |
You can read RFC 3261 at this Internet Engineering Task Force Web site.