A SIP Header Only Requires 6 Fields, But There Are Hundreds More


In networking communications, Session Initiation Protocol (SIP) is what helps us set up and carry out calls and other forms of messaging over the internet. It’s also a vital ingredient that makes Voice over Internet Protocol (VoIP) systems work.

For a session to begin, a caller must somehow get the attention of a receiver or recipient, and that starts by sending a SIP INVITE along with some header fields.

SIP overview graphic from Nextiva.

SIP INVITEs vs SIP Headers

SIP headers accompany SIP INVITEs, and these elements work in tandem. An INVITE sends a request message to set up a call, video, or chat session. Meanwhile, SIP headers convey precise information, instructions, and metadata regarding the message and its associated session.

SIP headers are like specialized tools in a toolbox, each designed for a specific purpose. Some fields convey information about the sender, others dictate the route a message will take, and others contribute to the overall quality of the conversation.

While there are hundreds of potential header fields, only a select few are required to set up a call. More specifically, there are six foundational headers that are essential for SIP to work properly.

What Are the Required SIP Header Fields?

Since SIP is a dynamic protocol, the exact number of fields can vary depending on the particular implementation and the requirements of the session. That’s because SIP is useful for multiple types of communication (like voice, video, and instant messaging) as well as specific configurations used by many of the best VoIP service providers.

This variability arises because different providers will implement different functionalities or have specific requirements for their systems. The inherent flexibility in SIP allows for easy customization and adaptation to many communication infrastructures.

Nevertheless, any and all SIP communication still needs to have the following SIP header fields to initiate a successful session:

  • From: Specifies the sender of the SIP request or response
  • To: Indicates the intended recipient of the SIP request or response
  • Call-ID: Generated by the sender, it’s a unique identifier that allows the related SIP messages to be grouped as part of the same call
  • CSeq: Contains a sequence number and method (such as INVITE, ACK, and BYE) to indicate the order of SIP messages in a transaction
  • Via: Provides information about the routing path and transport protocol used in the SIP message
  • Max-Forwards: Specifies the maximum number of hops a SIP message can pass through before being discarded
  • For non-technical folks, that’s a lot of mumbo jumbo to take in at once—but each field can be broken down into simpler parts with more digestible details.


    The From header field helps the recipient identify who started the SIP communication process, and it allows for proper routing and processing of the request or response.

    The information in the SIP header field usually includes the SIP address (also known as the SIP URI) of the sender, which comes in the format of “user@domain.”

    For example, a From header field may look like this:

    From: sip:[email protected]

    In this case, [email protected] represents the SIP address of the sender, with alice as the username and example.com as the domain.


    The To header field specifies the ultimate destination for the SIP message, allowing SIP entities to route and process it accordingly.

    It can contain a display name, a SIP or telephone URI, and additional parameters that are optional. Without this field, it wouldn’t be possible to route and handle SIP messages correctly.

    An example of the To header field looks like this:

    To: Alice ;tag=abcd1234

    Here, Alice is the display name, and sip:[email protected] is the SIP URI representing the user Alice at the domain example.com.

    Next, the ;tag=abcd1234 part is an optional parameter for transaction-matching and ensuring message integrity.


    The Call-ID header is a unique identifier that helps ensure all SIP messages related to the same call are correctly associated and processed.

    The specific Call-ID value is generated by the initiating user agent (UA) and remains constant throughout the entire session. It is typically a string of alphanumeric characters, and its uniqueness is crucial for proper call handling and tracking.

    A standard Call-ID header field looks like this:

    Call-ID: [email protected]

    In this super imaginative example, 1234567890 is the unique identifier for the call, and example.com represents the domain where the call occurs.

    By utilizing the Call-ID header field, SIP entities can track and correlate all SIP messages within a session, including the initial call setup, subsequent message exchanges, and the termination at the end of the call.


    The C-Seq (Command Sequence) header field is meant for organizing messages within a particular transaction. By using this header field, SIP entities can ensure that requests and responses are processed correctly and in the intended order.

    The C-Seq header field contains two components: a sequence number and a method. The sequence number is a numeric value that starts at one and changes in increments for each request within a transaction. The method specifies the type of SIP request being sent.

    Here’s an example of a C-Seq header field:

    CSeq: 1 INVITE

    Here, the 1 represents the sequence number, and INVITE indicates that the SIP request is an invitation to establish a session.

    It’s important to note that the C-Seq header field is specific to each transaction and does not carry over between different transactions. Therefore, the sequence number is always unique within one given transaction but may be reused in subsequent transactions.


    The Via header field provides information about the routing path and transport protocol used in SIP messages. It helps ensure the SIP message is correctly delivered across different networks and through various intermediaries.

    An example of the text in a Via field is as follows:

    Via: SIP/2.0/UDP;branch=z9hG4bK74bf9

    In this example, SIP/2.0/UDP indicates the protocol and version being used, while represents the IP address and port number of the intermediary SIP server. In this case, the branch parameter is a unique identifier the sender generates to help track the SIP message in the communication flow.

    Remember that the syntax here is essential, as correctly formatting the Via field ensures that intermediary servers can accurately process and route SIP messages. This format typically includes the protocol/version, IP address, port number, and any additional parameters separated by semicolons.


    If you’ve ever tried to make a call that timed out or had a message that refused to be sent from your VoIP phone, it might have been because you reached the Max-Forwards limit.

    The Max-Forwards header field puts a cap on the number of times a SIP message can be forwarded on its path to the recipient. This field contains a numerical value that decreases with each hop. When the value reaches zero, the message is discarded.


    Max-Forwards: 5

    Here, the initial value of the Max-Forwards field is set to five. As the SIP message travels through different proxies or intermediaries, each entity decreases the value by one before forwarding the message. If the value goes from five to zero at any point, the message is dropped, and no further forwarding occurs.

    By setting an appropriate initial value for the Max-Forwards field, network administrators can control the maximum number of hops a SIP message can make. This ensures efficient routing, prevents message duplication, and avoids potential network congestion caused by loops or excessive forwarding.

    Other Important SIP Header Fields

    In addition to the six mandatory header fields, the SIP protocol allows for extensibility—meaning that new fields can be defined to meet the needs of specific applications or scenarios. As a result, the possible number of header fields is practically unlimited, with over a hundred more commonly used SIP header fields.

    Of these hundred, a few of the most important or noteworthy SIP headers are:

    • Content-Type: A header that conveys the precise nature of the data contained within the message body. Whether audio, video, or text, it ensures that the recipient understands the format and can process it accordingly.
    • Contact: This header helps ensure messages are correctly delivered to the intended recipient in a SIP-based system. It contains contact information such as an IP address or a URI (Uniform Resource Identifier), allowing other participants to connect with the intended recipient.
    • User-Agent: A header that identifies the software or device responsible for sending or receiving the SIP message. This information is vital in troubleshooting and ensuring compatibility between different systems.
    • Route: Specifies the path that a SIP message should follow from the sender to the recipient. It contains a list of network elements (like routers and proxies) that the message should pass through before reaching its destination. This header helps in routing messages across complex networks.
    • Authorization: Ensures that only certain users or devices can access SIP-based services and is typically used during the registration or authentication processes. It contains credentials, such as a username and password, that verify the user’s or device’s identity.

    By including these headers in SIP-based systems, it’s possible to improve compatibility between different systems and enable important functionalities that make communication smooth and efficient. Thus, if you need to create specific protocols to enhance the VoIP phone services at your own business, look to these header fields for help.

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