SIP messages and media protocols

This section provides an overview of SIP messages and how they communicate information about SIP sessions and how SDP, RTP, and RTCP fits in with SIP communications.

SIP uses clear text messages to start, maintain, and end media sessions between SIP user agent clients (UACs) and user agent servers (UASs). These messages form a SIP dialog. A typical SIP dialog begins with an INVITE request message sent from a UAC to another UAC or to a UAS. The first INVITE request message attempts to start a SIP call and includes information about the sending UAC and the receiving UAC as well as information about the communication session.

If only two UACs are involved as shown below, the receiving UAC (Phone B) responds with a 180 Ringing and then a 200 OK SIP response message that informs Phone A that Phone B received and accepted the request. Phone A then sends an ACK message to notify Phone B that the SIP response was received. Phone A and Phone B can then participate in the RTP media session set up by the SIP messages.

When the phone call is complete, one of the UACs (in the example Phone B) hangs up sending a BYE request message to Phone A. Phone A then sends a 200 OK response to Phone B acknowledging that the session has ended.

Basic SIP dialog between two UACs

SIP Phone A (Sending UAC PhoneA@10.31.101.20)

SIP Phone B (Receiving UAC PhoneB@10.31.101.30)

1. INVITE (SIP request message to invite SIP Phone B to start a SIP session).
2. 180 Ringing (SIP ringing response to the INVITE request).
3. 200 OK (SIP response to the INVITE request to inform SIP Phone A

that the request is accepted).

4. ACK (SIP request message to confirm that

SIP Phone A received the response from SIP Phone B).

5. RTP Media session between Phone Aand Phone B.
6. BYE (SIP request message from SIP Phone B to end the SIP session).
7. 200 OK (SIP response to the BYE request to end the SIP session).

If a UAS in the form of a SIP proxy server is involved, similar messages are sent and received, but the proxy server participates as an intermediary in the initial call setup. In the example below the SIP proxy server receives the INVITE request from Phone A and forwards it to Phone B. The proxy server then sends a 100 Trying response to Phone A. Phone B receives the INVITE request and responds with a 180 Ringing and then a 200 OK SIP response message. These messages are received by the proxy server and forwarded to Phone A to notify Phone A that Phone B received and accepted the request. Phone A then sends an ACK message to notify Phone B that the SIP response was received. This response is received by the proxy server and forwarded to Phone B. Phone A and Phone B can then participate in the media session independently of the proxy server.

When the phone call is complete Phone B hangs up sending a BYE request message to Phone A. Phone A then sends a 200 OK response to Phone B acknowledging that the session has ended.

Basic SIP dialog between UACs with a SIP proxy server UAS

2. INVITE (Forwarded by the UAS to Phone B.)
3. 180 Ringing (SIP ringing response to the INVITE request.)
4. 200 OK (SIP response to the INVITE request to inform Phone A that the request is accepted.)
5. BYE (SIP request message from Phone B to end the SIP session.)

The SIP messages include SIP headers that contain names and addresses of Phone A, Phone B and the proxy server. This addressing information is used by the UACs and the proxy server during the call set up.

The SIP message body includes Session Description Protocol (SDP) statements that Phone A and Phone B use to establish the media session. The SDP statements specify the type of media stream to use for the session (for example, audio for SIP phone calls) and the protocol to use for the media stream (usually the Real Time Protocol (RTP) media streaming protocol).

Hardware accelerated RTP processing

Phone A includes the media session settings that it would like to use for the session in the INVITE message. Phone B includes its response to these media settings in the 200 OK response. Phone A’s ACK response confirms the settings that Phone A and Phone B then use for the media session.

Hardware accelerated RTP processing

FortiGates can offload RTP packet processing to network processor (NP) interfaces. This acceleration greatly enhances the overall throughput and resulting in near speed RTP performance.

SIP request messages

SIP sessions always start with a SIP request message (also just called a SIP request). SIP request messages also establish, maintain, and terminate SIP communication sessions. The following table lists some common SIP request message types.

Common SIP request message types

Message Type	Description
INVITE	A client sends an INVITE request to invite another client to participate in a multimedia session. The INVITE request body usually contains the description of the session.
ACK	The originator of an INVITE message sends an ACK request to confirm that the final response to an INVITE request was received. If the INVITE request did not contain the session description, it must be included in the ACK request.
PRACK	In some cases, SIP uses provisional response messages to report on the progress of the response to a SIP request message. The provisional response messages are sent before the final SIP response message. Similar to an ACK request message, a PRACK request message is sent to acknowledge that a provisional response message has been received.
OPTIONS	The UA uses OPTIONS messages to get information about the capabilities of a SIP proxy. The SIP proxy server replies with a description of the SIP methods, session description protocols, and message encoding that are supported.
BYE	A client sends a BYE request to end a session. A BYE request from either end of the SIP session terminates the session.
CANCEL	A client sends a CANCEL request to cancel a previous INVITE request. A CANCEL request has no effect if the SIP server processing the INVITE sends a final response to the INVITE before receiving the CANCEL.

 

response messages

Message Type	Description
REGISTER	A client sends a REGISTER request to a SIP registrar server with information about the current location (IP address and so on) of the client. A SIP registrar server saves the information it receives in REGISTER requests and makes this information available to any SIP client or server attempting to locate the client.
Info	For distributing mid-session signaling information along the signaling path for a SIP call. I
Subscribe	For requesting the current state and state updates of a remote node.
Notify	Informs clients and servers of changes in state in the SIP network.
Refer	Refers the recipient (identified by the Request-URI) to a third party according to the contact information in the request.
Update	Opens a pinhole for new or updated SDP information.
Response codes (1xx, 202, 2xx, 3xx, 4xx, 5xx, 6xx)	Indicates the status of a transaction. For example: 200 OK, 202 Accepted, or 400 Bad Request.

SIP response messages

SIP response messages (often just called SIP responses) provide status information in response to SIP request messages. All SIP response messages include a response code and a reason phrase. There are five SIP response message classes. They are described below.

There are also two types of SIP response messages, provisional and final. Final response messages convey the result of the request processing, and are sent reliably. Provisional responses provide information on the progress of the request processing, but may not be sent reliably. Provisional response messages start with 1xx and are also called informational response messages.

Informational (or provisional)

Informational or provisional responses indicate that a request message was received and imply that the endpoint is going to process the request. Information messages may not be sent reliably and may not require an acknowledgment.

If the SIP implementation uses Provisional Response Acknowledgment (PRACK) (RFC 3262) then informational or provisional messages are sent reliably and require a PRACK message to acknowledge that they have been received.

Informational responses can contain the following reason codes and reason phrases:

100 Trying

• Ringing
• Call is being forwarded
• Queued

SIP response messages

• Session progress

Success

Success responses indicate that a request message was received, understood, and accepted. Success responses can contain the following reason codes and reason phrases:

200 OK

202 Accepted

Redirection

Redirection responses indicate that more information is required for the endpoint to respond to a request message. Redirection responses can contain the following reason codes and reason phrases:

• Multiple choices
• Moved permanently
• Moved temporarily

305 Use proxy

380 Alternative service

Client error

Client error responses indicate that a request message was received by a server that contains syntax that the server cannot understand (i.e. contains a syntax error) or cannot comply with. Client error responses include the following reason codes and reason phrases:

400 Bad request               401 Unauthorized

402 Payment required          403 Forbidden

404 Not found                405 Method not allowed

406 Not acceptable            407 Proxy authentication required

408 Request time-out          409 Conflict

410 Gone                      411 Length required

413 Request entity too large  414 Request-URL too large

415 Unsupported media type   420 Bad extension

• Temporarily not available
• Call leg/transaction does not exist
• Loop detected

484 Address incomplete        483 Too many hops

486 Busy here                 485 Ambiguous

488 Not acceptable here       487 Request canceled

Server error

Server error responses indicate that a server was unable to respond to a valid request message. Server error responses include the following reason codes and reason phrases:

• Server internal error
• Not implemented
• Bad gateway

502 Service unavailable

• Gateway time-out
• SIP version not supported

message start line

Global failure

Global failure responses indicate that there are no servers available that can respond to a request message. Global failure responses include the following reason codes and reason phrases:

600 Busy everywhere

• Decline
• Does not exist anywhere 606 Not acceptable

SIP message start line

The first line in a SIP message is called the start line. The start line in a request message is called the requestline and the start line in a response message is called the status-line.

Request-line

The first line of a SIP request message. The request-line includes the SIP message type, the SIP protocol version, and a Request URI that indicates the user or service to which this request is being addressed. The following example request-line specifies the INVITE message type, the address of the sender of the message (inviter@example.com), and the SIP version: INVITE sip:inviter@example.com SIP/2.0

Status-line

The first line of a SIP response message. The status-line includes the SIP protocol version, the response code, and the reason phrase. The example status-line includes the SIP version, the response code (200) and the reason phrase (OK). SIP/2.0 200 OK

SIP headers

Following the start line, SIP messages contain SIP headers (also called SIP fields) that convey message attributes and to modify message meaning. SIP headers are similar to HTTP header fields and always have the following format:

<header_name>:<value>

SIP messages can include the SIP headers listed in the following table:

SIP headers

SIP headers

SIP Header	Description
Allow	Lists the set of SIP methods supported by the UA generating the message. All methods, including ACK and CANCEL, understood by the UA MUST be included in the list of methods in the Allow header field, when present. For example: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE
Call-ID	A globally unique identifier for the call, generated by the combination of a random string and the sender’s host name or IP address. The combination of the To, From, and Call-ID headers completely defines a peer-to-peer SIP relationship between the sender and the receiver. This relationship is called a SIP dialog. Call-ID: ddeg45e793@10.31.101.30
Contact	Included in SIP request messages, the Contact header contains the SIP URI of the sender of the SIP request message. The receiver uses this URI to contact the sender. For example: Contact: Sender <sip:sender@10.31.100.20>t
Content-Length	The number of bytes in the message body (in bytes). Content-Length: 126
Content-Type	In addition to SIP headers, SIP messages include a message body that contains information about the content or communication being managed by the SIP session. The Content-Type header specifies what the content of the SIP message is. For example, if you are using SIP with SDP, the content of the SIP message is SDP code. Content-Type: application/sdp
CSeq	The command sequence header contains a sequence integer that is increased for each new SIP request message (but is not incremented in the response message). This header also includes the request name found in the request message requestline. For example: CSeq: 1 INVITE
Expires	Gives the relative time after which the message (or content) expires. The actual time and how the header is used depends on the SIP method. For example: Expires: 5
From	Identifies the sender of the message. Responses to a message are sent to the address of the sender. The following example includes the sender’s name (Sender) and the sender’s SIP address (sender@10.31.101.20.): From: Sender <sip:sender@10.31.101.20>

 

headers

SIP Header	Description
Max-forwards	An integer in the range 0-255 that limits the number of proxies or gateways that can forward the request message to the next downstream server. Also called the number of hops, this value is decreased every time the message is forwarded. This can also be useful when the client is attempting to trace a request chain that appears to be failing or looping in mid-chain. For example: Max-Forwards: 30
P-AssertedIdentity	The P-Asserted-Identity header is used among trusted SIP entities to carry the identity of the user sending a SIP message as it was verified by authentication. See RFC 3325. The header contains a SIP URI and an optional display-name, for example: P-Asserted-Identity: “Example Person” <sip:10.31.101.50>
RAck	Sent in a PRACK request to support reliability of information or provisional response messages. It contains two numbers and a method tag. For example: RAck: 776656 1 INVITE
Record-Route	Inserted into request messages by a SIP proxy to force future requests to be routed through the proxy. In the following example, the host at IP address 10.31.101.50 is a SIP proxy. The lr parameter indicates the URI of a SIP proxy in Record-Route headers. Record-Route: <sip:10.31.101.50;lr>
Route	Forces routing for a request message through one or more SIP proxies. The following example includes two SIP proxies: Route: <sip:172.20.120.10;lr>, <sip:10.31.101.50;lr>
RSeq	The RSeq header is used in information or provisional response messages to support reliability of informational response messages. The header contains a single numeric value. For example: RSeq: 33456
To	Identifies the receiver of the message. The address in this field is used to send the message to the receiver. The following example includes the receiver’s name (Receiver) and the receiver’s SIP address (receiver@10.31.101.30.): To: Receiver <sip:receiver@10.31.101.30>
Via	Indicates the SIP version and protocol to be used for the SIP session and the address to which to send the response to the message that contains the Via field. The following example Via field indicates to use SIP version 2, UDP for media communications, and to send the response to 10.31.101.20 using port 5060. Via: SIP/2.0/UDP 10.31.101.20:5060

30

The SIP message body and SDP session profiles

The SIP message body and SDP session profiles

The SIP message body describes the session to be initiated. For example, in a SIP phone call the body usually includes audio codec types, sampling rates, server IP addresses and so on. For other types of SIP session the body could contain text or binary data of any type which relates in some way to the session. The message body is included in request and response messages.

Two possible SIP message body types:

l Session Description Protocol (SDP), most commonly used for SIP VoIP. l Multipurpose Internet Mail Extensions (MIME)

SDP is most often used for VoIP and FortiGates support SDP content in SIP message bodies. SDP is a textbased protocol used by SIP to control media sessions. SDP does not deliver media but provides a session profile that contains media details, transport addresses, parameter negotiation, and other session description metadata for the participants in a media session. The participants use the information in the session profile to negotiate how to communicate and to manage the media session. SDP is described by RFC 4566.

An SDP session profile always contains session information and may contain media information. Session information appears at the start of the session profile and media information (using the m= attribute) follows.

SDP session profiles can include the attributes listed in the following table.

SDP session profile attributes

Attribute	Description
a=	Attributes to extend SDP in the form a=<attribute> or a=<attribute>:<value>.
b=	Contains information about the bandwidth required for the session or media in the form b=<bandwidth_type>:<bandwidth>.
c=	Connection data about the session including the network type (usually IN for Internet), address type (IPv4 or IPv6), the connection source address, and other optional information. For example: c=IN IPv4 10.31.101.20
i=	A text string that contains information about the session. For example: i=A audio presentation about SIP
k=	Can be used to convey encryption keys over a secure and trusted channel. For example: k=clear:444gdduudjffdee

The SIP message body and SDP session profiles

Attribute	Description
m=	Media information, consisting of one or more lines all starting with m= and containing details about the media including the media type, the destination port or ports used by the media, the protocol used by the media, and a media format description. m=audio 49170 RTP 0 3 m-video 3345/2 udp 34 m-video 2910/2 RTP/AVP 3 56 Multiple media lines are needed if SIP is managing multiple types of media in one session (for example, separate audio and video streams). Multiple ports for a media stream are indicated using a slash. 3345/2 udp means UDP ports 3345 and 3346. Usually RTP uses even-numbered ports for data with the corresponding one-higher odd ports used for the RTCP session belonging to the RTP session. So 2910/2 RTP/AVP means ports 2910 and 2912 are used for RTP and 2911 and 2913 are used for RTCP. Media types include udp for an unspecified protocol that uses UDP, RTP or RTP/AVP for standard RTP and RTP/SAVP for secure RTP.
o=	The sender’s username, a session identifier, a session version number, the network type (usually IN for Internet), the address type (for example, IPv4 or IPv6), and the sending device’s IP address. The o= field becomes a universal identifier for this version of this session description. For example: o=PhoneA 5462346 332134 IN IP4 10.31.101.20
r=	Repeat times for a session. Used if a session will be repeated at one or more timed intervals. Not normally used for VoIP calls. The times can be in different formats. For example: r=7d 1h 0 25h r=604800 3600 0 90000
s=	Any text that describes the session or s= followed by a space. For example: s=Call from inviter
t=	The start and stop time of the session. Sessions with no time restrictions (most VoIP calls) have a start and stop time of 0. t=0 0
v=	SDP protocol version. The current SDP version is 0 so the v= field is always: v=0
z=	Time zone adjustments. Used for scheduling repeated sessions that span the time between changing from standard to daylight savings time. z=2882844526 -1h 2898848070 0

 

Example SIP messages

The following example SIP INVITE request message was sent by PhoneA to PhoneB. The first nine lines are the SIP headers. The SDP profile starts with v=0 and the media part of the session profile is the last line, starting with m=.

INVITE sip:PhoneB@172.20.120.30 SIP/2.0

Via: SIP/2.0/UDP 10.31.101.50:5060

From: PhoneA <sip:PhoneA@10.31.101.20>

To: PhoneB <sip:PhoneB@172.20.120.30>

Call-ID: 314159@10.31.101.20

CSeq: 1 INVITE

Contact: sip:PhoneA@10.31.101.20

Content-Type: application/sdp

Content-Length: 124 v=0

o=PhoneA 5462346 332134 IN IP4 10.31.101.20 s=Let’s Talk t=0 0

c=IN IP4 10.31.101.20 m=audio 49170 RTP 0 3

The following example shows a possible 200 OK SIP response message in response to the previous INVITE request message. The response includes 200 OK which indicates success, followed by an echo of the original SIP INVITE request followed by PhoneB’s SDP profile.

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.31.101.50:5060

From: PhoneA <sip:PhoneA@10.31.101.20>

To: PhoneB <sip:PhoneB@172.20.120.30>

Call-ID: 314159@10.31.101.20

CSeq: 1 INVITE

Contact: sip:PhoneB@10.31.101.30

Content-Type: application/sdp

Content-Length: 107 v=0 o=PhoneB 124333 67895 IN IP4 172.20.120.30 s=Hello! t=0 0

c=IN IP4 172.20.120.30 m=audio 3456 RTP 0

SIP can support multiple media streams for a single SIP session. Each media steam will have its own c= and m= lines in the body of the message. For example, the following message includes three media streams:

INVITE sip:PhoneB@172.20.120.30 SIP/2.0

Via: SIP/2.0/UDP 10.31.101.20:5060

From: PhoneA <sip:PhoneA@10.31.101.20>

To: PhoneB <sip:PhoneB@172.20.120.30>

Call-ID: 314159@10.31.101.20

CSeq: 1 INVITE

Contact: sip:PhoneA@10.31.101.20

Content-Type: application/sdp

Content-Length: 124 v=0

o=PhoneA 5462346 332134 IN IP4 10.31.101.20 s=Let’s Talk

Example SIP messages

t=0 0 c=IN IP4 10.31.101.20 m=audio 49170 RTP 0 3 c=IN IP4 10.31.101.20 m=audio 49172 RTP 0 3 c=IN IP4 10.31.101.20 m=audio 49174 RTP 0 3

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Common SIP VoIP configurations

This section describes some common SIP VoIP configurations and simplified SIP dialogs for these configurations. This section also shows some examples of how adding a FortiGate affects SIP processing.

Peer to peer configuration

In the peer to peer configuration shown below, two SIP phones (in the example, FortiFones) communicate directly with each other. The phones send SIP request and response messages back and forth between each other to establish the SIP session.

SIP peer to peer configuration

Peer to peer configurations are not very common because they require the SIP phones to keep track of the names and addresses of all of the other SIP phones that they can communicate with. In most cases a SIP proxy or redirect server maintains addresses of a large number of SIP phones and a SIP phone starts a call by contacting the SIP proxy server.

SIP proxy server configuration

A SIP proxy server act as intermediary between SIP phones and between SIP phones (for example, two FortiFones) and other SIP servers. As shown below, SIP phones send request and response messages the SIP proxy server. The proxy server forwards the messages to other clients or to other SIP proxy servers. Proxy servers can hide SIP phones by proxying the signaling messages. To the other users on the VoIP network, the signaling invitations look as if they come from the SIP proxy server.

redirect server configuration

SIP in proxy mode

SIP proxy server

A common SIP configuration would include multiple networks of SIP phones. Each of the networks would have its own SIP server. Each SIP server would proxy the communication between phones on its own network and between phones in different networks.

SIP redirect server configuration

A SIP redirect server accepts SIP requests, maps the addresses in the request into zero or more new addresses and returns those addresses to the client. The redirect server does not initiate SIP requests or accept calls. As shown below, SIP clients send INVITE requests to the redirect server, which then looks up the destination address. The redirect server returns the destination address to the client. The client uses this address to send the INVITE request directly to the destination SIP client.

Common SIP VoIP configurations                                                                                    SIP registrar configuration

SIP in redirect model

SIP redirect server

SIP registrar configuration

A SIP registrar accepts SIP REGISTER requests from SIP phones for the purpose of updating a location database with this contact information. This database can then become a SIP location service that can be used by SIP proxy severs and redirect servers to locate SIP clients. As shown below, SIP clients send REGISTER requests to the SIP registrar.

 

SIP registrar and proxy servers

SIP with a FortiGate

Depending on your security requirements and network configuration FortiGates may be in many different places in a SIP configuration. This section shows a few examples.

The diagram below shows a FortiGate installed between a SIP proxy server and SIP phones on the same network. The FortiGate is operating in transparent mode so both the proxy server and the phones are on the same subnet. In this configuration, called SIP inspection without address translation, the FortiGate could be protecting the SIP proxy server on the private network by implementing SIP security features for SIP sessions between the SIP phones and the SIP proxy server.

Common SIP VoIP configurations                                                                                             SIP with a FortiGate

SIP network with FortiGate in transparent mode

call by proxy server. the INVITE request to Phone B.

The phone rings.

The phones and server use the same SIP dialogs as they would if the FortiGate was not present. However, the FortiGate can be configured to control which devices on the network can connect to the SIP proxy server and can also protect the SIP proxy server from SIP vulnerabilities.

The following diagram shows a FortiGate operating in NAT/Route mode and installed between a private network and the Internet. Some SIP phones and the SIP proxy server are connected to the private network and some SIP phones are connected to the Internet. The SIP phones on the Internet can connect to the SIP proxy server through the FortiGate and communication between SIP phones on the private network and SIP phones on the Internet must pass through the FortiGate.

SIP network with FortiGate in NAT/Route mode

1. 1^{. SIP phone B registers with} SIP Phone B
2. SIP phone A registers with SIP proxy server

(PhoneB@172.20.120.30) SIP proxy server.       using the SIP proxy server virtual IP.

2. Phone A dials Phone B
by sending an INVITE request to the SIP proxy server.	3. The proxy server looks up the SIP address of Phone B and forwards	4. Phone B is notified of an incoming

the INVITE request to Phone B.      ^{call by proxy server} – phone rings.

5. RTP Media session opens between

Phone A and Phone B when Phone B answers

The phones and server use the same SIP dialog as they would if the FortiGate was not present. However, the FortiGate can be configured to control which devices on the network can connect to the SIP proxy server and can also protect the SIP proxy server from SIP vulnerabilities. In addition, the FortiGate has a firewall virtual IP that forwards packets sent to the SIP proxy server Internet IP address (172.20.120.50) to the SIP proxy server internal network IP address (10.31.101.30).

Since the FortiGate is operating in NAT/Route mode it must translate packet source and destination IP addresses (and optionally ports) as the sessions pass through the FortiGate. Also, the FortiGate must translate the addresses contained in the SIP headers and SDP body of the SIP messages. As well the FortiGate must open SIP and RTP pinholes through the FortiGate. SIP pinholes allow SIP signaling sessions to pass through the FortiGate between phones and between phones and SIP servers. RTP pinholes allow direct RTP communication between the SIP phones once the SIP dialog has established the SIP call. Pinholes are opened automatically by the FortiGate. Administrators do not add security policies for pinholes or for RTP sessions. All that is required is a security policy that accepts SIP traffic.

Opening an RTP pinhole means opening a port on a FortiGate interface to allow RTP traffic to use that port to pass through the FortiGate between the SIP phones on the Internet and SIP phones on the internal network. A pinhole only accepts packets from one RTP session. Since a SIP call involves at least two media streams (one from Phone A to Phone B and one from Phone B to Phone A) the FortiGate opens two RTP pinholes. Phone A sends RTP packets through a pinhole in port2 and Phone B sends RTP packets through a pinhole in port1. The FortiGate opens the pinholes when required by the SIP dialog and closes the pinholes when the SIP call is completed. The FortiGate opens new pinholes for each SIP call.

Each RTP pinhole actually includes two port numbers. The RTP port number as defined in the SIP message and an RTCP port number, which is the RTP port number plus 1. For example, if the SIP call used RTP port 3346 the FortiGate would create a pinhole for ports 3346 and 3347.

 

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Inside FortiOS: Voice over IP (VoIP) protection

The FortiOS SIP Application Layer Gateway (ALG) allows SIP calls to pass through a FortiGate by opening SIP and RTP pinholes and performing source and destination IP address and port translation for SIP and RTP packets.

There are a large number of SIP-related Internet Engineering Task Force (IETF) documents (Request for

Comments) that define behavior of SIP and related applications. FortiOS completly support RFC 3261 for SIP, RFC 4566 for SDP and RFC 3262 for Provisional Response Acknowledgment (PRACK). FortiOS also supports other SIP and SIP-related RFCs and performs Deep SIP message inspection for SIP statements defined in other SIP RFCs.

Advanced voice over IP protection

The FortiOS SIP Application Level Gateway (ALG) protects Voice over IP (SIP and SDP) services in Unified Communication and NGN/IMS networks with the following advanced VoIP defense mechanisms.

Deep SIP message inspection (also called deep SIP header inspection)

Verifies SIP and SDP header syntax and protects SIP servers from potential SIP Fuzzing attacks. When a violation is detected, FortiOS can impose counter measures and can also send automatic SIP response messages to offload processing from the SIP server.

SIP message rate limiting

Allows rate limiting of SIP messages per SIP request method. This prevents a SIP server from overload or from DoS attacks using particular SIP methods. For example, FortiOS can protect SIP servers from a flood of SIP REGISTER or INVITE messages, which can be caused by a DoS attack or a flash crowd.

RTP and RTCP pinholing

RTP pinholing only forwards RTP/RTCP packets that conform to the particular session description of the associated SIP dialog. If a SIP dialog is finished, FortiOS automatically closes the pinhole. RTP/RTCP pinholing is supported by FortiASIC acceleration and achieves high packet throughput at low jitter and delay.

Stateful SIP dialog tracking

FortiOS tracks SIP message sequences and prevents unwanted SIP messages that are not related to a particular SIP dialog. For instance, FortiOS can detect malicious SIP BYE messages that do not conform with the associated context of the SIP dialog.

Inspecting SIP over SSL/TLS (secure SIP)

Some SIP phones and SIP servers use SSL or TLS to encrypt SIP signalling traffic. To allow SIP over SSL/TLS calls to pass through the FortiGate unit, the encrypted signalling traffic has to be unencrypted and inspected. FortiOS intercepts and unencrypts and inspects the SIP packets. Allowed packets are then re-encrypted and forwarded to their destination.

Carrier grade

Inspecting SIP on multiple ports

FortiOS can detect and inspect SIP and SDP UDP and TCP sessions and SIP SSL sessions and ou can configure the ports that the SIP ALG monitors for these sessions. In addition you can configure two different ports for SIP UDP sessions and two different ports for SIP TCP sessions. The port configuration can be changed without affecting other parts of the SIP configuration.

Carrier grade protection

To protect VoIP infrastructure in carrier networks, FortiOS complies with typical carrier requirements for availability and robustness.

High availability

FortiOS supports a hot failover configuration with an active and a standby FortiGate device. FortiOS dynamically updates the context on the standby unit with SIP and RTP related data. This enables the standby unit to takeover stable voice calls in case of a planned or unplanned outage or failover of the active unit.

Geographical redundancy of SIP servers

In FortiOS SIP server cluster configurations the active and standby units can be deployed in different geographical locations. This configuration prevents a total outage of a SIP server infrastructure if one location goes offline. FortiOS supports the detection of SIP server outages (loss of heartbeats) and a redirect of SIP messages to the redundant SIP server location.

Logging and Reporting

FortiOS can log call related information internally or to an external SYSLOG or FortiAnalyzer unit. This includes event logs that show particular SIP-related attacks or syntax violations with SIP messages or logs that summarize call statistics.

NAT/NAPT

FortiOS performs configurable network address translation for IP addresses in the SIP and SDP header. The SIP ALG follows the configured NAT addresses in firewall virtual IPs and changes SIP header IP addresses accordingly. RTP NAT is controlled by SIP/SDP and the firewall policy. This allows translating an unlimited number of IP addresses without adding specific RTP policies.

Header manipulation

FortiOS SIP and SDP header manipulation supports SIP Network Address Translation (NAT) through FortiGate units configured as NAT firewalls.

NAT/NAPT

Hosted NAT traversal (HNT)

In many service provider networks, CPE firewall devices provide NAT without application awareness. This causes issues for SIP/SDP and RTP traffic, since UAC IP address information references to the internal network behind the far end firewall. VoIP calls cannot be connected successfully. FortiOS mitigates far end NAT issues (called Hosted NAT traversal) by probing the first RTP packet from the UAC and learning the far end NA(P)T binding.

FortiOS then updates the internal NAT binding for RTP accordingly.

IP address conservation for NAT

In case of SIP and RTP NAT IP the original address information can get lost after translating to the provisioned IP addresses. This IP address information is sometimes required for detailed billing records or debugging purposes. FortiOS can maintain the original IP address information in a translated SIP header by adding it to the SIP/SDP info line (i=) or by adding it to the original attribute (o=). Either option can be selected depending on the SIP billing environment.

SIP ALG activation

The FortiOS SIP ALG is applied to SIP traffic accepted by a firewall policy that includes a VoIP profile. The VoIP profile controls how the SIP ALG processes SIP sessions. FortiOS also includes a high-performance SIP session helper that provides limited SIP functionality. In most cases the SIP ALG should be used because the SIP ALG supports the complete range of FortiOS SIP features.

 

	• IP routing and forwarding • IPsec VPN encryption, decryption
	• Rate limiting and message blocking • Stateful SIP tracking • Message, header, and SDP syntax checking • Network surveillance • NAT and IP topology Hiding • Logging and debugging
	• Intrusion detection and prevention • Defined by Fortinet and enterprise signatures • SIP decoder identifies SIP sessions
	• Security policy • IPsec VPN encryption, decryption • Access control
	• Native (D)DoS prevention • Anomaly detection and prevention

SIP over IPv6

FortiOS, operating in NAT/Route and in transparent mode supports SIP over IPv6. The SIP ALG can process SIP messages that use IPv6 addresses in the headers, bodies, and in the transport stack. The SIP ALG cannot modify the IPv6 addresses in the SIP headers so FortiGate units cannot perform SIP or RTP NAT over IPv6 and also cannot translate between IPv6 and IPv4 addresses.

Platform support and hardware acceleration

FortiOS supports VoIP protection with the SIP ALG on all FortiGate hardware platforms. Whenever a FortiGate unit provides FortiASIC or SPM HW acceleration, the SIP ALG will use this option to fast-path RTP/RTCP traffic.

As well, since the SIP ALG is proxy-based, SIP control packets are not offloaded to NP4 or NP6 processors. But actual voice or other media traffic can be offloaded to NP4 or NP6 processors after the SIP session is established. Many FortiGate units also support low latencey hardware acceleration configurations that also enhance SIP voice transmission.

FortiGate hardware acceleration provides a high throughput solution at very low jitter and delay. FortiOS provides efficient and highly scalable protection for VoIP in emerging Enterprise and Carrier network. This complements Fortinet’s NGFW and UTM offerings. VoIP protection can be easily added to any firewall policy just by adding a VoIP profile.

Platform support and hardware acceleration

VoIP protection is supported in FortiAnalyzer and FortiManager. Centralized logging and management are essential for carrier and MSSP service provider and are influencing business case calculations.

 

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