freeswitch/libs/sofia-sip/libsofia-sip-ua/docs/conformance.docs

/*  -*- html -*- */
/**

@page sofia_sip_conformance SIP and SDP Protocol Features in Sofia-SIP

This document describes how Sofia-SIP stack supports specifications listed
below.

<table border=0>
<tr valign="top">
<td>
<a href="#3261">RFC 3261</a> <br>
<a href="#2617">RFC 2617</a> <br>
<a href="#3262">RFC 3262</a> <br>
<a href="#3263">RFC 3263</a> <br>
<a href="#3265">RFC 3265</a> <br>
<a href="#2806">RFC 2806</a> <br>
<a href="#2976">RFC 2976</a> <br>
<a href="#3311">RFC 3311</a> <br>
<a href="#3313">RFC 3313</a> <br>
<a href="#3323">RFC 3323</a> <br>
<a href="#3326">RFC 3326</a> <br>
<a href="#3325">RFC 3325</a> <br>
<a href="#3327">RFC 3327</a> <br>
</td><td>
<a href="#3329">RFC 3329</a> <br>
<a href="#3361">RFC 3361</a> <br>
<a href="#3420">RFC 3420</a> <br>
<a href="#3428">RFC 3428</a> <br>
<a href="#3486">RFC 3486</a> <br>
<a href="#3515">RFC 3515</a> <br>
<a href="#3581">RFC 3581</a> <br>
<a href="#3608">RFC 3608</a> <br>
<a href="#3680">RFC 3680</a> <br>
<a href="#3824">RFC 3824</a> <br>
<a href="#3840">RFC 3840</a> <br>
<a href="#3841">RFC 3841</a> <br>
<a href="#3842">RFC 3842</a> <br>
</td><td>
<a href="#3856">RFC 3856</a> <br>
<a href="#3857">RFC 3857</a> <br>
<a href="#3858">RFC 3858</a> <br>
<a href="#3859">RFC 3859</a> <br>
<a href="#3860">RFC 3860</a> <br>
<a href="#3891">RFC 3891</a> <br>
<a href="#3892">RFC 3892</a> <br>
<a href="#3903">RFC 3903</a> <br>
<a href="#4028">RFC 4028</a> <br>
<a href="#4168">RFC 4168</a> <br>
<a href="#4320">RFC 4320</a> <br>
<a href="#4488">RFC 4488</a> <br>
<a href="#5057">RFC 5057</a> <br>
</td><td>
<a href="#4566">RFC 4566</a> <br>
<a href="#2327">RFC 2327</a> <br>
<a href="#3264">RFC 3264</a> <br>
<a href="#3266">RFC 3266</a> <br>
<a href="#3312">RFC 3312</a> <br>
<a href="#3388">RFC 3388</a> <br>
<a href="#3407">RFC 3407</a> <br>
<a href="#3524">RFC 3524</a> <br>
<a href="#3551">RFC 3551</a> <br>
<a href="#3556">RFC 3556</a> <br>
<a href="#3605">RFC 3605</a> <br>
<a href="#3890">RFC 3890</a> <br>
</td></tr>
</table>

<table border=1 cellpadding=4 cellspacing=0>
<tr>
   <th>Feature</th>
   <th>Supported</td>
   <th>Notes</td>
</tr>

<a name="3261"></a>
<tr valign=top>
    <th align="left">
	@RFC3261: Basic SIP Protocol
    </th>
    <td>
	The SIP registration and dialog level implementation enables the
	application to operate as a SIP UA, SIP proxy or a redirect server
	according to the @RFC3261.

	The @RFC3261 functionality is divided into five layers:
	-# <a href="sip/index.html">message syntax and encoding</a>
	-# <a href="tport/index.html">transport</a>
	-# <a href="nta/index.html">transaction</a>
	-# transaction user (UAS and UAC cores, proxy core)
	-# SIP elements: <a href="nua/index.html">user agent</a>
           client and server, proxies,
	   registrars
    </td>
    <td>
        &nbsp;
    </td>
</tr>

<a name="3261.19"></a> <a name="3261.20"></a>
<tr valign=top>
    <th align="left">
	@RFC3261 Sections&nbsp;19&nbsp;and&nbsp;20:<br>
	Syntax and encoding
    </td>
    <td>
	The supported @RFC3261 methods are: @b REGISTER, @b OPTIONS, @b
	INVITE, @b ACK, @b CANCEL, @b BYE, as well as extension methods
	<a href="#2976"><b>INFO</b></a>,
	<a href="#3262"><b>PRACK</b></a>,
        <a href="#3265"><b>SUBSCRIBE</b></a>,
        <a href="#3265"><b>NOTIFY</b></a>,
        <a href="#3311"><b>UPDATE</b></a>,
        <a href="#3428"><b>MESSAGE</b></a>,
        <a href="#3515"><b>REFER</b></a>,
	and
	<a href="#3903"><b>PUBLISH</b></a>.

	Sofia-SIP supports the following SIP headers as specified in
        @RFC3261 or its extensions (generating, parsing and syntax
        checking):

	@ref sip_accept "Accept",
	@ref sip_accept_encoding "Accept-Encoding",
	@ref sip_accept_language "Accept-Language",
	@ref sip_alert_info "Alert-Info" (extension in @VERSION_1_12_7),
	@ref sip_allow "Allow",
	@ref sip_authentication_info "Authentication-Info",
	@ref sip_authorization "Authorization",
	@ref sip_call_id "Call-ID" ("i"),
	@ref sip_call_info "Call-Info",
	@ref sip_contact "Contact" ("m"),
	@ref sip_content_disposition "Content-Disposition",
        @ref sip_content_encoding "Content-Encoding" ("e"),
	@ref sip_content_language "Content-Language",
	@ref sip_content_length "Content-Length" ("l"),
	@ref sip_content_type "Content-Type" ("c"),
	@ref sip_cseq "CSeq",
	@ref sip_date "Date",
	@ref sip_error_info "Error-Info",
	@ref sip_expires "Expires",
	@ref sip_from "From" ("f"),
	@ref sip_in_reply_to "In-Reply-To",
	@ref sip_max_forwards "Max-Forwards",
	@ref sip_min_expires "Min-Expires",
	@ref sip_mime_version "MIME-Version",
	@ref sip_organization "Organization",
	@ref sip_p_asserted_identity "P-Asserted-Identity"
        (extension in @VERSION_1_12_7),
	@ref sip_p_preferred_identity "P-Preferred-Identity"
        (extension in @VERSION_1_12_7),
	@ref sip_priority "Priority",
	@ref sip_proxy_authenticate "Proxy-Authenticate",
	@ref sip_proxy_authorization "Proxy-Authorization",
	@ref sip_proxy_require "Proxy-Require",
	@ref sip_record_route "Record-Route",
	@ref sip_refer_sub "Refer-Sub" (@VERSION_1_12_5),
	@ref sip_remote_party_id "Remote-Party-ID" (extension in @VERSION_1_12_7),
	@ref sip_reply_to "Reply-To" (extension in @VERSION_1_12_7),
	@ref sip_require "Require",
	@ref sip_retry_after "Retry-After",
	@ref sip_route "Route",
	@ref sip_server "Server",
	@ref sip_subject "Subject" ("s"),
	@ref sip_supported "Supported" ("k"),
	@ref sip_timestamp "Timestamp",
	@ref sip_to "To" ("t"),
	@ref sip_unsupported "Unsupported",
	@ref sip_user_agent "User-Agent",
	@ref sip_via "Via" ("v"),
	@ref sip_warning "Warning", and
	@ref sip_www_authenticate "WWW-Authenticate".

	Unknown headers (extension headers) are supported and can be passed
	to/received from application as name-value pairs.

	It is possible to extend SIP parser in run-time with header-specific
        parsers.
    </td>
    <td>
	- Automatic escaping of reserved characters has not been
	  implemented.
	- Using NUL (zero byte) in double-quoted strings has not been implemented
   </td>
</tr>

<a name="3261.18"></a>
<tr valign=top>
    <th align="left">
	@RFC3261 Section 18:<br>
	UDP and TCP transports
    </th>
    <td>
        UDP and TCP on both IP4 and IP6 are supported.

        The UDP size limit of 1300 bytes is enforced by default. If limit is
        exceeded, TCP is tried instead. If TCP connection is refused, UDP is
        tried if message size is less than 64 kilobytes. Limit is adjustable
        via parameter NTATAG_UDP_MTU().

	TCP connections are reused by client. However, server closes
	connections after idle time of 30 minutes (by default). The idle
	time limit is adjustable with TPTAG_IDLE() (given as argument to
	nta_agent_add_tport() or nta_agent_create()).

	Server tries to use same TCP connection to return response as the
	request was received.

	Only one SIP message is accepted per UDP message, as per @RFC3261.
    </td>
    <td>
	There is <a href="#4168">experimental support for SCTP</a>, too.
    </td>
</tr>

<a name="3261.17"></a>
<tr valign=top>
    <th align="left">
	@RFC3261 Section 17:<br> Transactions
    </th>
    <td>
        Transaction state engines function as specified in @RFC3261 section
	17. There is special handling of methods @b INVITE, @b ACK, and @b
	CANCEL. There are two modes for transaction state engines,
	User-Agent and Proxy modes.

	Default values for SIP timers are those specified by @RFC3261. The
	values for T1, T1x64, T2 and T4 can be changed via
        configuration tags defined in <sofia-sip/nta_tag.h>.

	The SIP timer C is implemented from @VERSION_1_12_7. Also, its value
	can be changed via configuration tag NTATAG_TIMER_C() defined in
	<sofia-sip/nta_tag.h>.
    </td>
    <td>
	&nbsp;
    </td>
</tr>


<a name="3261.26"></a>
<tr valign=top>
    <th align="left">
	@RFC3261 Section 26:<br> Security
    </th>
    <td>
        TLS and SIPS URIs has been implemented. Currently, TLS does not
	require certificate from client nor it does check it if one is
	provided.
    </td>
    <td>
	Missing:
	- Authorizing connections with TLS certificates
	- S/MIME
    </td>
</tr>

<a name="2617"></a>
<tr valign=top>
    <th align="left">
	@RFC2617: HTTP Digest Authentication
    </th>
    <td>
	Sofia-SIP includes authentication client and server modules
	implementing HTTP Digest authentication.

	HTTP Digest is a simple challenge-response authentication
	scheme defined in @RFC2617 based on the UA sending a checksum
	calculated over specific values in response to a challenge sent by
	the server (proxy or UA).

        Checksum calculation supports MD5 (@RFC1321). The algorithm for
        calculating MD5 digest hash can be MD5, MD5sess or be
        @RFC2069-compatible algorithm. The quality-of-protection (qop)
        parameters "auth", "auth-int" and none (missing) are supported. The
        "opaque" parameter is supported.

	The SIP authentication headers supported (generating, parsing and
	syntax checking) are:
	@ref sip_authorization "Authorization",
	@ref sip_authentication_info "Authentication-Info",
	@ref sip_proxy_authenticate "Proxy-Authenticate",
	@ref sip_proxy_authentication_info "Proxy-Authentication-Info",
	@ref sip_proxy_authorization "Proxy-Authorization", and
	@ref sip_www_authenticate "WWW-Authenticate".

	SIP interface to the modules is implemented as defined in @RFC3261
	(sections 8.1.3.5, 22.2, 22.3, 22.4).

	An @RFC2617 header
	@ref sip_proxy_authentication_info "Proxy-Authentication-Info"
        is not listed in @RFC3261 but it is nevertheless supported by
        Sofia-SIP.
    </td>
    <td>
	Missing:
        - Using nextnonce
	- Mutual authentication
    </td>
</tr>

<a name="3262"></a>
<tr valign=top>
    <th align="left">
	@RFC3262: PRACK and 100rel
    </th>
    <td>
	PRACK method is supported within dialog as defined in RFC3262.
        Semantics of reliable provisional responses are supported:
        - including 100rel Required header in provisional responses if
          request had 100rel
	- generation of PRACK based on 100rel option tag in Require header of
	  a provisional response, and
        - automatic re-transmission of provisional responses.

	The SIP headers supported (generating, parsing and syntax checking)
	are @ref sip_rseq "RSeq" and @ref sip_rack "RAck".
    </td>
    <td>
	&nbsp;
    </td>
</tr>

<a name="3263"></a>
<tr valign=top>
    <th align="left">
	@RFC3263: Locating SIP Servers
    </th>
    <td>
	Support for SIP server address resolution from SIP or SIPS URI using
	NAPTR, SRV, A or AAAA records in DNS as defined in @RFC3263.
    </td>
    <td>
	- Resolving any other types of URIs than SIP or SIPS URIs, e.g., IM:
	or PRES: URIs.
    </td>
</tr>

<a name="3265"></a>
<tr valign=top>
    <th align="left">
	@RFC3265: SIP Event Notifications
    </th>
    <td>
	SIP extensions for subscribing and processing asynchronous event
	notifications as defined in @RFC3265.

	Includes dialog level support for sending and refreshing SUBSCRIBE
	and receiving NOTIFY messages.

	The SIP headers explicitly supported (generating, parsing and
	syntax checking) are
        @ref sip_event "Event" ("o"),
	@ref sip_allow_events "Allow-Events", and
        @ref sip_subscription_state "Subscription-State"

	Note: currently there is no support for forked SUBSCRIBE requests.
    </td>
    <td>
	Application must take care of:
	- Subscribing, generating or processing specific event types
          and interpreting the content of event data is up to application
    </td>
</tr>

<a name="2806"></a>
<tr valign=top>
    <th align="left">
	@RFC2806: tel URI
    </th>
    <td>
	Sofia-SIP supports handling of any URI type. Sofia-SIP parses tel:
        URIs.
    </td>
    <td>
	Missing:
	- Resolving the tel: URIs
    </td>
</tr>

<a name="2976"></a>
<tr valign=top>
    <th align="left">
	@RFC2976: INFO
    </th>
    <td>
	INFO method is supported within a dialog natively.
    </td>
    <td>
	Not implemented:
	- Generating or processing contents of INFO requests
    </td>
</tr>


<a name="3311"></a>
<tr valign=top>
    <th align="left">
	@RFC3311: UPDATE
    </th>
    <td>
	UPDATE method as defined in RFC3311. UPDATE allows a client to
	update parameters of a session (such as the set of media streams and
	their codecs) even within early dialogs.

	Offer-Answer negotiation with UPDATE is implemented in nua.
    </td>
    <td>
	Application must take care of:
        - Initiating UPDATE requests
    </td>
</tr>


<a name="3313"></a>
<tr valign=top>
    <th align="left">
	@RFC3313: Media Authentication
    </th>
    <td>
	Sofia-SIP provides <a href="#3261.19">generic support</a> for
	extension headers and parameters. P-Media-Authorization header is
	supported as an @ref sip_unknown "extension header".
    </td>
    <td>
        Application must take care of:
	- Passing the authorization token to QoS reservation request
    </td>
</tr>


<a name="3323"></a>
<tr valign=top>
    <th align="left">
	@RFC3323: Privacy
    </th>
    <td>
	@ref sip_privacy "Privacy" header is supported (generating, parsing
        and syntax checking).

	Call-Id header is generated from cryptographically random id without
	host name or IP address. By default, @ref sip_contact "Contact" and
	@ref sip_via "Via" headers contain only IP address that can be
	dynamically allocated.

	Application can enter any SIP URI and display name to From header,
	e.g., @code Anonymous <sip:anonymous@example.org> @endcode.
    </td>
    <td>
        Application must take care of:
	- Properly populating the URIs and display names within SIP headers
	- Not including any optional headers that could reveal identity
	- Generating of Privacy header with appropriate values
	- Generating of Proxy-Require: privacy
	- Using anonymous callback URIs etc.
    </td>
</tr>


<a name="3326"></a>
<tr valign=top>
    <th align="left">
	@RFC3326: Reason
    </th>
    <td>
	Sofia-SIP supports @ref sip_reason "Reason" header (generating,
	parsing and syntax checking).
    </td>
    <td>
        Application must take care of:
	- Generating or processing Reason headers
    </td>
</tr>


<a name="3325"></a>
<tr valign=top>
    <th align="left">
	@RFC3325: Asserted Identity
    </th>
    <td>
	Sofia-SIP supports
	@ref sip_p_asserted_identity "P-Asserted-Identity" and
	@ref sip_p_preferred_identity "P-Preferred-Identity" headers
        (generating, parsing and syntax checking). Also the non-standard
        header @ref sip_remote_party_id "Remote-Party-ID" is supported.

        @NEW_1_12_7.
    </td>
    <td>
	Not implemented:
        - <a href="#3323">id privacy</a>
	- Recognizing trust domains and enforcing handling of headers
	  based on those
    </td>
</tr>

<a name="3327"></a>
<tr valign=top>
    <th align="left">
	@RFC3327: Path
    </th>
    <td>
	User-agent engine can add "path" option tag to Supported header of
	REGISTER requests.

	Sofia-SIP explicitly supports @ref sip_path "Path" header
	(generating, parsing and syntax checking).
    </td>
    <td>
	&nbsp;
    </td>
</tr>

<a name="3329"></a>
<tr valign=top>
    <th align="left">
	@RFC3329:
	Security Agreement
    </th>
    <td>
	Some support of the SIP Security Mechanism Agreement procedures. The
	client is able to insert Security-Client and Security-Verify header
	with fake @e d-ver value.

	Sofia-SIP explicitly supports (generating, parsing
	and syntax checking)
	@ref sip_security_client "Security-Client",
	@ref sip_security_server "Security-Server", and
	@ref sip_security_verify "Security-Verify" headers.

	Security-mechanism supported is "digest".
    </td>
    <td>
	Correct @e d-ver value is not calculated.
    </td>
</tr>

<a name="3361"></a>
<tr valign=top>
    <th align="left">
	@RFC3361: DHCPv4 option for locating SIP servers.
    </th>
    <td>
	Sofia-SIP supports outbound proxy.
    </td>
    <td>
	Application must take care of:
	- passing outbound proxy name or address from DHCP client
          to Sofia-SIP stack.
    </td>
</tr>

<a name="3420"></a>
<tr valign=top>
    <th align="left">
	@RFC3420: message/sipfrag
    </th>
    <td>
	Sofia-SIP passes the received SIP message headers to application
	which can create a message/sipfrag payload.
    </td>
    <td>
        Application must take care of:
	- processing the SIP message fragments
    </td>
</tr>

<a name="3428"></a>
<tr valign=top>
    <th align="left">
	@RFC3428: MESSAGE
    </th>
    <td>
	MESSAGE method is supported natively.
    </td>
    <td>
	&nbsp;
    </td>
</tr>

<a name="3486"></a>
<tr valign=top>
    <th align="left">
	@RFC3486: Compressing SIP
    </th>
    <td>
	Sofia-SIP provides support for using comp=sigcomp parameters in @ref
	sip_via "Via" header and @ref url_t "SIP URIs", indicating
	support for compression.
    </td>
    <td>
	SigComp itself is not supported.
    </td>
</tr>

<a name="3515"></a>
<tr valign=top>
    <th align="left">
	@RFC3515: REFER
    </th>
    <td>
	REFER method is supported natively. Sofia-SIP processes incoming
        REFER requests and generates NOTIFY with correct
	@ref sip_event "Event" header automatically.

	Further notifications can be automatically generated actions when
	nua_invite() is given referrer's nua handle in NUTAG_NOTIFY_REFER().

	Sofia-SIP explicitly supports (generating, parsing and syntax
	checking) @ref sip_refer_to "Refer-To" ("r") SIP header.

	See also support for
	<a href="#3891">RFC 3891</a> and
        <a href="#3892">RFC 3892</a>.
    </td>
    <td>
        &nbsp;
    </td>
</tr>

<a name="3608"></a>
<tr valign=top>
    <th align="left">
	@RFC3608: Service-Route
    </th>
    <td>
	Sofia-SIP supports @ref sip_service_route "Service-Route" that can
	be used to provide a mechanism by which a registrar may inform a
	registering UA of a service route. User-Agent will optionally use
	the route provided in @ref sip_service_route "Service-Route" header.

	The SIP header explicitly supported (generating, parsing and
	syntax checking) is @ref sip_service_route "Service-Route".
    </td>
    <td>
	&nbsp;
    </td>
</tr>

<a name="3680"></a>
<tr valign=top>
    <th align="left">
	@RFC3680: "reg" event
    </th>
    <td>
	Sofia-SIP supports <a href="#3265">generic SIP event support</a> for
	subscribing SIP event packages and receiving notifications for them.
	Subscriptions are refreshed before expiration when needed and
	subscriptions are terminated on request. Sofia-SIP takes care of
        notified subscription states.

	UA can SUBSCRIBE to a registration state event package after sending
	initial REGISTER to, e.g., 3GPP network and use it to follow its
        registration status.
    </td>
    <td>
	Application must take care of:
	- Generating subscriptions for "reg" event
	- Processing notifications for "reg" event
	  - Handling of XML document in notifications
    </td>
</tr>

<a name="3824"></a>
<tr valign=top>
    <th align="left">
	@RFC3824: ENUM
    </th>
    <td>
	Tel URIs are supported in any headers including URI parameters,
	e.g., To, From, Contact headers, and Request-URI of the outgoing SIP
	request provided that the next hop is given as SIP or SIPS URI.
    </td>
    <td>
	Stack can not resolve E.164 number to address of next
	hop. A proxy in the network must resolve E.164 numbers with ENUM.
    </td>
</tr>

<a name="3840"></a>
<tr valign=top>
    <th align="left">
	@RFC3840: Callee Capabilities
    </th>
    <td>
	Feature parameters can be added to SIP profiles and sent within
	Contact header of REGISTER request as header parameters.

	UAC can optionally generate media tags for Contact header using
	local media profile.

	Feature parameters can also be sent within any other SIP request as
	extension parameters of Contact header.
    </td>
    <td>
	Application must take care of:
        - Processing the feature parameters received in the Contact header
    </td>
</tr>


<a name="3841"></a>
<tr valign=top>
    <th align="left">
	@RFC3841: Caller Preferences
    </th>
    <td>
	Built-in support for user-agent behavior.

	UAC can optionally generate Accept-Contact header using local media
	profile.

	SIP parser parses the Accept-Contact and Reject-Contact headers.

        ACK and CANCEL request messages sent have same values
	for Accept-Contact/Reject-Contact or Request-Disposition
	headers as the original request had.

	There are functions for calculating score for contacts.

	The SIP headers explicitly supported (generating,
	parsing and syntax checking) are:
	@ref sip_request_disposition "Request-Disposition" ("d"),
	@ref sip_accept_contact "Accept-Contact" ("a"),
	@ref sip_reject_contact "Reject-Contact" ("j"),
    </td>
    <td>
	Application must take care of:
        - UAS processing incoming Accept-Contact or Reject-Contact headers
    </td>
</tr>

<a name="3842"></a>
<tr valign=top>
    <th align="left">
	@RFC3842: Message waiting event
    </th>
    <td>
	Sofia-SIP supports <a href="#3265">generic SIP event support</a> for
	subscribing SIP event packages and receiving notifications for them.
	Subscriptions are refreshed before expiration when needed and
	subscriptions are terminated on request. Sofia-SIP takes care of
        notified subscription states.
    </td>
    <td>
	Application must take care of:
	- Generating subscriptions for "message-summary" event
          - Including correct @ref sip_event "Event" and
            @ref sip_accept "Accept" headers in the request (if needed)
	- Processing notifications for "message-summary" event
	  - Handling of summary information in notifications
    </td>
</tr>

<a name="3856"></a><a name="3859"></a>
<tr valign=top>
    <th align="left">
	@RFC3856: Presence <br>
	@RFC3859: Common Profile for Presence
    </th>
    <td>
	Sofia-SIP supports <a href="#3265">generic SIP event support</a> for
	subscribing SIP event packages and receiving notifications for them.
	Subscriptions are refreshed before expiration when needed and
	subscriptions are terminated on request. Sofia-SIP takes care of
        notified subscription states.

	Note: Usage of pres: URI is supported only if the next hop URI to
	where to send SUBSCRIBE is a SIP URI (e.g. of outbound proxy).
	Resolving of pres: URIs by DNS is not supported.
    </td>
    <td>
	Application must take care of:
	- Generating subscriptions for "presence" event
          - Including correct @ref sip_event "Event" and
            @ref sip_accept "Accept" headers in the request (if needed)
	- Processing notifications for "presence" event
	  - Handling of presence information in notifications
    </td>
</tr>

<a name="3857"></a> <a name="3858"></a>
<tr valign=top>
    <th align="left">
	@RFC3857: "winfo" event template package<br>
	@RFC3858: winfo format
    </th>
    <td>
	Sofia-SIP supports <a href="#3265">generic SIP event support</a> for
	subscribing SIP event packages and receiving notifications for them.
	Subscriptions are refreshed before expiration when needed and
	subscriptions are terminated on request. Sofia-SIP takes care of
        notified subscription states.
    </td>
    <td>
        Application must take care of:
	- Generating subscriptions for winfo events
          - Including correct @ref sip_event "Event" and
            @ref sip_accept "Accept" headers in the request (if needed)
	- Processing notifications for winfo events:
	  - Processing watcherinfo XML documents
    </td>
</tr>

<a name="3860"></a>
<tr valign=top>
    <th align="left">
	@RFC3860: Common Profile for IM
    </th>
    <td>
	Sofia-SIP supports handling of any URI type. Sofia-SIP parses "im:"
        URIs.
    </td>
    <td>
        Application must take care of:
	- resolving the "im:" URI
    </td>
</tr>

<a name="3891"></a>
<tr valign=top>
    <th align="left">
	@RFC3891: Replaces
    </th>
    <td>
	@ref sip_replaces "Replaces" header is explicitly supported
	(generating, parsing and syntax checking).
    </td>
    <td>
        Application must take care of:
	- generating @ref sip_replaces "Replaces" header from a dialog and
          matching a dialog with a Replaces header received
    </td>
</tr>

<a name="3892"></a>
<tr valign=top>
    <th align="left">
	@RFC3892: Referred-By
    </th>
    <td>
	@ref sip_referred_by "Referred-By" header is explicitly supported
	(generating, parsing and syntax checking).

	Referred-by token can be sent and received in
	text-based SIP message body.
    </td>
    <td>
        Application must take care of:
	- Generating or processing @ref sip_referred_by "Referred-By" headers
	- Generating (and encrypting) or verifying (and decrypting) of
	  Referred-by tokens
    </td>
</tr>

<a name="3903"></a>
<tr valign=top>
    <th align="left">
	@RFC3903: PUBLISH
    </th>
    <td>
	PUBLISH method is supported natively. The SIP headers
	explicitly supported (generating, parsing and syntax checking) are
	@ref sip_etag "SIP-ETag" and @ref sip_if_match "SIP-If-Match".

	The <a href="nua/index.html">user-agent engine</a> keep published
	data refreshed until nua_unpublish() is called.
    </td>
    <td>
        Application must take care of:
        - Including correct @ref sip_event "Event" in the request
        - Permanently storing @SIPETag
    </td>
</tr>

<a name="4028"></a>
<tr valign=top>
    <th align="left">
	@RFC4028: Session Timers
    </th>
    <td>
	The SIP session-timer ("timer") extension is supported.

        The session-expires value and refreshing party is negotiated in
	<a href="nua/index.html">user-agent engine</a>. When user-agent
	engine is responsible for refreshes, it will initiate re-INVITE or
	UPDATE transaction at regular intervals.

	If there has been no SIP activity in session during the refresh
	period, it will try to automatically terminate the call by sending a
	@b BYE request.

        The SIP headers explicitly supported (generating, parsing and
	syntax checking) are @ref sip_session_expires "Session-Expires" ("x") and
	@ref sip_min_se "Min-SE".
    </td>
    <td>
	&nbsp;
    </td>
</tr>

<a name="4168"></a>
<tr valign=top>
    <th align="left">
	@RFC4168: SCTP as Transport for SIP
    </th>
    <td>
	The transport=sctp URI parameter is supported. The SCTP transport
	protocol is supported as experimental. It is enabled with
	configure script argument --enable-sctp.

	The framing of SIP messages over SCTP is not specified clearly in
	@RFC4168. It is possible to send SIP messages smaller than 64K over
	SCTP.
    </td>
    <td>
	&nbsp;
    </td>
</tr>


<a name="4320"></a>
<tr valign=top>
    <th align="left">
	@RFC4320: Actions Addressing Identified Issues with SIP's Non-INVITE Transaction
    </th>
    <td>
	The action 1 (make the best use of provisional responses) is
	supported when NTATAG_EXTRA_100(1) is used with nua_create() or
	nta_agent_create(). The 100 Trying provisional response is sent
	after T2 is expired or when a retransmission is received after T2/2
        after the initial request.

	The action 2 (remove the useless late-response storm) is supported
	by default. The 408 timeout response is not forwarded by default (it's
	forwarding can be enabled with NTATAG_PASS_408(1), however).
    </td>
    <td>
        Application must include NTATAG_EXTRA_100(1) with nua_create() or
	nta_agent_create() tags.
    </td>
</tr>

<a name="4488"></a>
<tr valign=top>
    <th align="left">
	@RFC4488: Suppression of REFER Implicit Subscription
    </th>
    <td>
	Sofia-SIP supports @ref sip_refer_sub "Refer-Sub" header
	(generating, parsing and syntax checking).

        The implicit subscription is suppressed by @nua, if the @ReferSub:
        true header is included in the REFER
	request (@ref nua_refer "on server side") or
	response (@ref nua_i_refer "on client side").

	@NEW_1_12_5
    </td>
    <td>
        The REFER client application must include SIPTAG_REFER_SUB_STR("true")
	in the nua_refer() tags.
    </td>
</tr>

<a name="5057"></a>
<tr valign=top>
    <th align="left">
	@RFC5057: Multiple Dialog Usages in SIP
    </th>
    <td>
	Sofia-SIP provides function sip_response_terminates_dialog() that
	can be used to determine the effect of error response with dialog.

	The nua UA engine uses sip_response_terminates_dialog().
    </td>
    <td>
        The client application must either use NUA or
        sip_response_terminates_dialog().
    </td>
</tr>

</table>

<table border=1 cellpadding=4 cellspacing=0>
<tr>
   <th>Feature</th>
   <th>Supported</td>
   <th>Notes</td>
</tr>

<a name="4566"></a>
<a name="2327"></a>
<a name="3266"></a>
<tr valign=top>
    <th align="left" align="left">
	@RFC4566: SDP: Session Description Protocol
    </th>
    <td>
	Generic support (generating, parsing and syntax checking) for SDP.
	The SDP
        @ref sdp_version_t "v=",
        @ref sdp_origin_t "o=",
	@ref sdp_connection_t "c=",
	@ref sdp_bandwidth_t "b=",
	@ref sdp_time_t "t=",
	@ref sdp_repeat_t "r=",
	@ref sdp_zone_t "z=",
	@ref sdp_key_t "k=",
	@ref sdp_attribute_t "a=", and
	@ref sdp_media_t "m="
        lines are separated and parsed. The "e=", "p=", "s=", and "i=" lines
	are separated.

	The attributes "a=sendrecv", "a=sendonly", "a=recvonly",
	"a=inactive", @ref sdp_rtpmap_s "a=rtpmap", and "a=fmtp" are parsed.

	The implementation partially implements @RFC4566. Note that
	definition of 'token' was updated in @RFC4566 and the parser has not
	been updated yet.
    </td>
    <td>
      @RFC4566 obsoletes
      - @RFC2327: SDP (Session Description Protocol)
      - @RFC3266: IP6 in SDP
    </td>
</tr>

<a name="3264"></a>
<tr valign=top>
    <th align="left">
	@RFC3264: SDP Offer/Answer Negotiation
    </th>
    <td>
	Generating and processing offers or answers.
    </td>
    <td>
	- "a=fmtp" parameters are not taken into account
          when generating or processing answer
    </td>
</tr>

<a name="3312"></a>
<tr valign=top>
    <th align="left">
	@RFC3312: Preconditions
    </th>
    <td>
	Sofia-SIP provides <a href="#2616">generic support</a> for attribute
	lines that conform to SDP syntax.

        Sofia-SIP supports handling SIP feature tags in
	@ref sip_proxy_require "Proxy-Require",
        @ref sip_require "Require",
        @ref sip_supported "Supported" ("k"),
        and
        @ref sip_unsupported "Unsupported" header.
    </td>
    <td>
	Application must take care of:
	- Semantics and handling of preconditions
	- Reservation of resources
    </td>
</tr>

<a name="3388"></a>
<tr valign=top>
    <th align="left">
	@RFC3388: Grouping of Media Lines
    </th>
    <td>
	Sofia-SIP provides <a href="#2616">generic support</a> for attribute
	lines that conform to SDP syntax.

	@RFC3388 defines mid, group, LS and FID are predefined strings to be
	used within attribute line
    </td>
    <td>
        Application must take care of:
	- Generating or processing the attribute lines
	- Grouping the media for transport accordingly
    </td>
</tr>


<a name="3407"></a>
<tr valign=top>
    <th align="left">
	@RFC3407: Capability Declaration
    </th>
    <td>
	Sofia-SIP provides <a href="#2616">generic support</a> for attribute
	lines that conform to SDP syntax.
    </td>
    <td>
        Application must take care of:
	- Defining sqn, cdsc, cpar etc. strings needed in a= line
	- Generating or processing the attribute lines
	- Capability negotiation itself
    </td>
</tr>


<a name="3524"></a>
<tr valign=top>
    <th align="left">
	@RFC3524: SRF
    </th>
    <td>
	Sofia-SIP provides <a href="#2616">generic support</a> for attribute
	lines that conform to SDP syntax.
    </td>
    <td>
        Application must take care of:
	- Defining SRF string needed in a= line
	- Generating or processing the attribute lines
    </td>
</tr>


<a name="3551"></a>
<tr valign=top>
    <th align="left">
	@RFC3551: RTP/AVP
    </th>
    <td>
	Sofia-SIP recognizes the RTP payload types for well-known audio and
	video codecs defined in @RFC3551.
    </td>
    <td>
        Application must take care of:
	- Audio or video processing
        - Generating a=rtpmap or a=fmtp lines when needed
    </td>
</tr>

<a name="3556"></a>
<tr valign=top>
    <th align="left">
	@RFC3556: Bandwidth
    </th>
    <td>
	Sofia-SIP provides <a href="#2616">generic support</a> for attribute
	lines that conform to SDP syntax.
    </td>
    <td>
        Application must take care of:
	- Generating or processing RS and RR bandwidth modifiers
	- Semantics of bandwidth allocation
    </td>
</tr>

<a name="3605"></a>
<tr valign=top>
    <th align="left">
	@RFC3605: RTCP attribute
    </th>
    <td>
	Sofia-SIP provides <a href="#2616">generic support</a> for attribute
	lines that conform to SDP syntax.
    </td>
    <td>
        Application must take care of:
	- Discovering port numbers
	- Generating or processing the RTCP attribute lines
    </td>
</tr>

<a name="3890"></a>
<tr valign=top>
    <th align="left">
	@RFC3890: TIAS
    </th>
    <td>
	Sofia-SIP provides <a href="#2616">generic support</a> for attribute
	lines that conform to SDP syntax.
    </td>
    <td>
        Application must take care of:
	- Generating or processing TIAS bandwidth modifiers
	- Generating or processing the maxprate attribute lines
    </td>
</tr>

</table>


*/

/* Overflow:

<a name="4566"></a>
<tr valign=top>
    <th align="left">
	@RFC4566: SDP: Session Description Protocol
    </th>
    <td>
    Obsoletes RFC2327 and RFC3266.
    </td>
    <td>
	&nbsp;
    </td>
</tr>


<a name="3320"></a>
<tr valign=top>
    <th align="left">
	@RFC3320: SigComp
    </th>
    <td>
    	Support for using SigComp for compression and
	decompression of SIP/SDP messages. By default, dynamic
	compression is used.

	Decompression using UDVM
    </td>
    <td>
	&nbsp;
    </td>
</tr>

<a name="3321"></a>
<tr valign=top>
    <th align="left">
	@RFC3321: SigComp Extended operations
    </th>
    <td>
	Support for SigComp extended operations.
    </td>
    <td>
	- Explicit Acknowledgment Scheme
	- Shared Compression
	- Checkpoint State
	- Implicit Deletion for Dictionary Update
    </td>
</tr>

<a name="3325"></a>
<tr valign=top>
    <th align="left">
	@RFC3325: Asserted Identity
    </th>
    <td>
	Explicit support (generating, parsing and syntax checking) for the
	following SIP headers: P-Asserted-Identity, P-Preferred-Identity
    </td>
    <td>
	- Recognizing trust domains and enforcing handling of headers
	  based on those
    </td>
</tr>

<a name="3485"></a>
<tr valign=top>
    <th align="left">
	@RFC3485: SIP/SDP Dictionary
    </th>
    <td>
	Support for SigComp static compression using SIP/SDP
	dictionary.
    </td>
    <td>
	&nbsp;
    </td>
</tr>




	- Implicitly registered user identities

<a name="3959"></a>
<tr valign=top>
    <th align="left">
	@RFC3959: early-session
    </th>
    <td>
	Early-session value can be used within Content-Disposition,
	Supported and Require headers.
    </td>
    <td>
	- Generating of Content-Disposition, Supported and Require
	- Handling of multipart bodies with early and session SDP
    </td>
</tr>

*/