A SIP alias is an alternative address for a SIP URI, which is the standard format for identifying a user or a service in a SIP network. A SIP URI consists of a user part, an at sign (@), and a host part, such as user@host. The user part can be a username, a phone number, or any other identifier. The host part can be a domain name, an IP address, or any other network address. For example, alice@example.com, +1234567890@voip.com, and bob@192.168.1.1 are valid SIP URIs1.
A SIP alias is a way of creating multiple SIP URIs for the same user or service, using different user parts or host parts. For example, if a user has a SIP URI of john.doe@cisco.com, he can also have a SIP alias of jdoe@cisco.com, or john.doe@voip.cisco.com, or any other combination of user and host parts. A SIP alias can be useful for providing different contact options, simplifying dialing, or hiding the real SIP URI for privacy or security reasons2.
To configure a SIP alias, you need to have a SIP account with a SIP service provider that supports SIP alias, such as iptel.org. You also need to have a SIP device or application that can register and use the SIP alias, such as a VoIP phone, a softphone, or a router. You can create and manage your SIP alias using the web interface of your SIP service provider, or using the configuration settings of your SIP device or application2.
The correct format for a SIP alias is the same as the format for a SIP URI, which is user@host. Therefore, the option C. John.doe@cisco.com is the correct format for a SIPalias, as it has a user part (John.doe) and a host part (cisco.com) separated by an at sign (@). The other options are not correct formats for a SIP alias, because:
A. 212-555-1212 : This is a phone number, not a SIP URI. It does not have a user part or a host part, and it does not have an at sign (@).
B. +12125551212 : This is also a phone number, not a SIP URI. It does not have a user part or a host part, and it does not have an at sign (@).
D. John.doe : This is only a user part, not a SIP URI. It does not have a host part, and it does not have an at sign (@).
References := Session Initiation Protocol (SIP) Parameters, How to configure SIP Alias | DrayTek
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