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janus_sip.c
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/*! \file janus_sip.c
* \author Lorenzo Miniero <lorenzo@meetecho.com>
* \copyright GNU General Public License v3
* \brief Janus SIP plugin
* \details Check the \ref sip for more details.
*
* \ingroup plugins
* \ref plugins
*
* \page sip SIP plugin documentation
* This is a simple SIP plugin for Janus, allowing WebRTC peers
* to register at a SIP server (e.g., Asterisk) and call SIP user agents
* through a Janus instance. Specifically, when attaching to the plugin peers
* are requested to provide their SIP server credentials, i.e., the address
* of the SIP server and their username/secret. This results in the plugin
* registering at the SIP server and acting as a SIP client on behalf of
* the web peer. Most of the SIP states and lifetime are masked by the plugin,
* and only the relevant events (e.g., INVITEs and BYEs) and functionality
* (call, hangup) are made available to the web peer: peers can call
* extensions at the SIP server or wait for incoming INVITEs, and during
* a call they can send DTMF tones. Calls can do plain RTP or SDES-SRTP.
*
* The concept behind this plugin is to allow different web pages associated
* to the same peer, and hence the same SIP user, to attach to the plugin
* at the same time and yet just do a SIP REGISTER once. The same should
* apply for calls: while an incoming call would be notified to all the
* web UIs associated to the peer, only one would be able to pick up and
* answer, in pretty much the same way as SIP forking works but without the
* need to fork in the same place. This specific functionality, though, has
* not been implemented as of yet.
*
* \section sipapi SIP Plugin API
*
* All requests you can send in the SIP Plugin API are asynchronous,
* which means all responses (successes and errors) will be delivered
* as events with the same transaction.
*
* The supported requests are \c register , \c unregister , \c call ,
* \c accept, \c decline , \c info , \c message , \c dtmf_info ,
* \c subscribe , \c unsubscribe , \c transfer , \c recording ,
* \c hold , \c unhold , \c update and \c hangup . \c register can be used,
* as the name suggests, to register a username at a SIP registrar to
* call and be called, while \c unregister unregisters it; \c call is used
* to send an INVITE to a different SIP URI through the plugin, while
* \c accept and \c decline are used to accept or reject the call in
* case one is invited instead of inviting; \c transfer takes care of
* attended and blind transfers (see \ref siptr for more details);
* \c hold and \c unhold can be used respectively to put a
* call on-hold and to resume it; \c info allows you to send a generic
* SIP INFO request, while \c dtmf_info is focused on using INFO for DTMF
* instead; \c message is the method you use to send a SIP message
* to the other peer; \c subscribe and \c unsubscribe are used to deal
* with SIP events, i.e., to send SUBSCRIBE requests that will result in
* NOTIFY asynchronous events; \c recording is used, instead, to record the
* conversation to one or more .mjr files (depending on the direction you
* want to record); \c update allows you to update an existing session
* (e.g., to do a renegotiation or force an ICE restart); finally, \c hangup
* can be used to terminate the communication at any time, either to
* hangup (BYE) an ongoing call or to cancel/decline (CANCEL/BYE) a call
* that hasn't started yet.
*
* No matter the request, an error response or event is always formatted
* like this:
*
\verbatim
{
"sip" : "event",
"error_code" : <numeric ID, check Macros below>,
"error" : "<error description as a string>"
}
\endverbatim
*
* Notice that the error syntax above refers to the plugin API messaging,
* and not SIP error codes obtained in response to SIP requests, which
* are notified using a different syntax:
*
\verbatim
{
"sip" : "event",
"result" : {
"event" : "<name of the error event>",
"code" : <SIP error code>,
"reason" : "<SIP error reason>",
"reason_header" : "<Reason header text; optional>",
"reason_header_protocol" : "<Reason header protocol; optional>",
"reason_header_cause" : "<Reason header cause code; optional>"
}
}
\endverbatim
*
* Coming to the available requests, you send a SIP REGISTER using the
* \c register request. To be more precise, a \c register request MAY result
* in a SIP REGISTER, as this method actually provides ways to start using
* a SIP account with no need for a registration. It is the case, for instance,
* of the so-called \c guest registrations: if you register as a \c guest ,
* it means you'll use the provided SIP URI in your \c From headers for calls,
* but you will actually not send a SIP REGISTER; this is especially useful
* for outgoing calls to services that don't require registration (e.g., IVR
* systems, or conference bridges), but also means you won't be able to
* receive calls unless peers know what your private SIP address is. A SIP
* REGISTER isn't sent also when registering as a \c helper : as we'll
* explain later, \c helper sessions are sessions only meant to facilitate
* the setup of \ref sipmc.
*
* That said, a \c register request has to be formatted as follows:
*
\verbatim
{
"request" : "register",
"type" : "<if guest or helper, no SIP REGISTER is actually sent; optional>",
"send_register" : <true|false; if false, no SIP REGISTER is actually sent; optional>,
"force_udp" : <true|false; if true, forces UDP for the SIP messaging; optional>,
"force_tcp" : <true|false; if true, forces TCP for the SIP messaging; optional>,
"sips" : <true|false; if true, configures a SIPS URI too when registering; optional>,
"rfc2543_cancel" : <true|false; if true, configures sip client to CANCEL pending INVITEs without having received a provisional response first; optional>,
"username" : "<SIP URI to register; mandatory>",
"secret" : "<password to use to register; optional>",
"ha1_secret" : "<prehashed password to use to register; optional>",
"authuser" : "<username to use to authenticate (overrides the one in the SIP URI); optional>",
"display_name" : "<display name to use when sending SIP REGISTER; optional>",
"user_agent" : "<user agent to use when sending SIP REGISTER; optional>",
"proxy" : "<server to register at; optional, as won't be needed in case the REGISTER is not goint to be sent (e.g., guests)>",
"outbound_proxy" : "<outbound proxy to use, if any; optional>",
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP REGISTER; optional>",
"contact_params" : "<array of key/value objects, to specify custom Contact URI params to add to the SIP REGISTER; optional>",
"incoming_header_prefixes" : "<array of strings, to specify custom (non-standard) headers to read on incoming SIP events; optional>",
"refresh" : "<true|false; if true, only uses the SIP REGISTER as an update and not a new registration; optional>",
"master_id" : "<ID of an already registered account, if this is an helper for multiple calls (more on that later); optional>",
"register_ttl" : "<integer; number of seconds after which the registration should expire; optional>"
}
\endverbatim
*
* A \c registering event will be sent back, as this is an asynchronous request.
*
* In case it is required to, this request will originate a SIP REGISTER to the
* specified server with the right credentials. 401 and 407 responses will be
* handled automatically, and so errors will not be notified back to the caller
* unless they're definitive (e.g., wrong credentials). A failure to register
* will return an error with name \c registration_failed. A successful registration,
* instead, is notified in a \c registered event formatted like this:
*
\verbatim
{
"sip" : "event",
"result" : {
"event" : "registered",
"username" : <SIP URI username>,
"register_sent" : <true|false, depending on whether a REGISTER was sent or not>,
"master_id" : <unique ID of this registered session in the plugin, if a potential master>
}
}
\endverbatim
*
* To unregister, just send an \c unregister request with no other arguments:
*
\verbatim
{
"request" : "unregister"
}
\endverbatim
*
* As before, an \c unregistering event will be sent back. Just as before,
* this will also send a SIP REGISTER in case it had been sent originally.
* A successful unregistration is notified in an \c unregistered event:
*
\verbatim
{
"sip" : "event",
"result" : {
"event" : "unregistered",
"username" : <SIP URI username>,
"register_sent" : <true|false, depending on whether a REGISTER was sent or not>
}
}
\endverbatim
*
* Once registered, you can call or wait to be called: notice that you won't
* be able to get incoming calls if you chose never to send a REGISTER at
* all, though.
*
* To send a SIP INVITE, you can use the \c call request, which has to
* be formatted like this:
*
\verbatim
{
"request" : "call",
"call_id" : "<user-defined value of Call-ID SIP header used in all SIP requests throughout the call; optional>",
"uri" : "<SIP URI to call; mandatory>",
"refer_id" : <in case this is the result of a REFER, the unique identifier that addresses it; optional>,
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP INVITE; optional>",
"srtp" : "<whether to mandate (sdes_mandatory) or offer (sdes_optional) SRTP support; optional>",
"srtp_profile" : "<SRTP profile to negotiate, in case SRTP is offered; optional>",
"secret" : "<password to use to call, only needed in case authentication is needed and no REGISTER was sent; optional>",
"ha1_secret" : "<prehashed password to use to call, only needed in case authentication is needed and no REGISTER was sent; optional>",
"authuser" : "<username to use to authenticate as to call, only needed in case authentication is needed and no REGISTER was sent; optional>",
"autoaccept_reinvites" : <true|false, whether we should blindly accept re-INVITEs with a 200 OK instead of relaying the SDP to the application; optional, TRUE by default>
}
\endverbatim
*
* A \c calling event will be sent back, as this is an asynchronous request.
*
* Notice that this request MUST be associated to a JSEP offer: there's no
* way to send an offerless INVITE via the SIP plugin. This will generate
* a SIP INVITE and send it according to the instructions. While a
* <code>100 Trying</code> will not be notified back to the user, a
* <code>180 Ringing</code> will, in a \c ringing event:
*
\verbatim
{
"sip" : "event",
"call_id" : "<value of SIP Call-ID header for related call>",
"result" : {
"event" : "ringing",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* If the call is declined, or any other error occurs, a \c hangup error
* event will be sent back. If the call is accepted, instead, an \c accepted
* event will be sent back to the user, along with the JSEP answer originated
* by the callee:
*
\verbatim
{
"sip" : "event",
"call_id" : "<value of SIP Call-ID header for related call>",
"result" : {
"event" : "accepted",
"username" : "<SIP URI of the callee>",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* At this point, PeerConnection-related considerations aside, the call
* can be considered established. A SIP ACK is sent automatically by the
* SIP plugin, so there's no action required of the application to do
* that manually.
*
* Notice that the SIP plugin supports early-media via \c 183 responses
* responses. In case a \c 183 response is received, it's sent back to
* the user, along with the JSEP answer originated by the callee, in
* a \c progress event:
*
\verbatim
{
"sip" : "event",
"call_id" : "<value of SIP Call-ID header for related call>",
"result" : {
"event" : "progress",
"username" : "<SIP URI of the callee>",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* In case the caller received a \c progress event, the following
* \c accepted event will NOT contain a JSEP answer, as the one received
* in the "Session Progress" event will act as the SDP answer for the session.
*
* Notice that you only use \c call to start a conversation, that is for
* the first INVITE. To update a session via a re-INVITE, e.g., to renegotiate
* a session to add/remove streams or force an ICE restart, you do NOT
* use \c call, but another request called \c update instead. This request
* needs no arguments, as the whole context is derived from the current
* state of the session. It does need the new JSEP offer to provide, though,
* as part of the renegotiation.
*
\verbatim
{
"request" : "update"
}
\endverbatim
*
* An \c updating event will be sent back, as this is an asynchronous request.
*
* While the \c call request allows you to send a SIP INVITE (and the
* \c update request allows you to update an existing session), there is
* a way to react to SIP INVITEs as well, that is to handle incoming calls.
* Incoming calls are notified to the application via \c incomingcall
* events:
*
\verbatim
{
"sip" : "event",
"call_id" : "<value of SIP Call-ID header for related call>",
"result" : {
"event" : "incomingcall",
"username" : "<SIP URI of the caller>",
"displayname" : "<display name of the caller, if available; optional>",
"callee" : "<SIP URI that was called (in case the user is associated with multiple public URIs)>",
"referred_by" : "<SIP URI header conveying the identity of the transferor, if this is a transfer; optional>",
"replaces" : "<call-ID of the call that this is supposed to replace, if this is an attended transfer; optional>",
"srtp" : "<whether the caller mandates (sdes_mandatory) or offers (sdes_optional) SRTP support; optional>",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* The \c incomingcall may or may not be accompanied by a JSEP offer, depending
* on whether the caller sent an offerless INVITE or a regular one. Either
* way, you can accept the incoming call with the \c accept request:
*
\verbatim
{
"request" : "accept",
"srtp" : "<whether to mandate (sdes_mandatory) or offer (sdes_optional) SRTP support; optional>",
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP OK; optional>"
"autoaccept_reinvites" : <true|false, whether we should blindly accept re-INVITEs with a 200 OK instead of relaying the SDP to the browser; optional, TRUE by default>
}
\endverbatim
*
* An \c accepting event will be sent back, as this is an asynchronous request.
*
* This will result in a <code>200 OK</code> to be sent back to the caller.
* An \c accept request must always be accompanied by a JSEP answer (if the
* \c incomingcall event contained an offer) or offer (in case it was an
* offerless INVITE). In the former case, an \c accepted event will be
* sent back just to confirm the call can be considered established;
* in the latter case, instead, an \c accepting event will be sent back
* instead, and an \c accepted event will only follow later, as soon as
* a JSEP answer is available in the SIP ACK the caller sent back.
*
* Notice that in case you get an incoming call while you're in another
* call, you will NOT get an \c incomingcall event, but a \c missed_call
* event instead, and just as a notification as there's no way to have
* two calls at the same time on the same handle in the SIP plugin:
*
\verbatim
{
"sip" : "event",
"call_id" : "<value of SIP Call-ID header for related call>",
"result" : {
"event" : "missed_call",
"caller" : "<SIP URI of the caller>",
"displayname" : "<display name of the caller, if available; optional>",
"callee" : "<SIP URI that was called (in case the user is associated with multiple public URIs)>"
}
}
\endverbatim
*
* Besides, you only use \c accept to answer the first INVITE. To accept a
* re-INVITE instead, which would be notified via an \c updatingcall event,
* you do NOT use \c accept, but the previously introduced \c update instead.
* This request needs no arguments, as the whole context is derived from the current
* state of the session. It does need the new JSEP answer to provide, though,
* as part of the renegotiation. As before, an \c updated event will be
* sent back, as this is an asynchronous request.
*
* Closing a session depends on the call state. If you have an incoming
* call that you don't want to accept, use the \c decline request; in all
* other cases, use the \c hangup request instead. Both requests need no
* additional arguments, as the whole context can be extracted from the
* current state of the session in the plugin:
*
\verbatim
{
"request" : "decline",
"code" : <SIP code to be sent, if not set, 486 is used; optional>",
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP request; optional>"
}
\endverbatim
*
\verbatim
{
"request" : "hangup",
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP BYE; optional>"
}
\endverbatim
*
* Since these are asynchronous requests, you'll get an event in response:
* \c declining if you used \c decline and \c hangingup if you used \c hangup.
*
* As anticipated before, when a call is declined or being hung up, a
* \c hangup event is sent instead, which is basically a SIP error event
* notification as it includes the \c code and \c reason . A regular BYE,
* for instance, would be notified with \c 200 and <code>SIP BYE</code>,
* although a more verbose description may be provided as well.
*
* When a session has been established, there are different requests that
* you can use to interact with the session.
*
* First of all, you can put a call on-hold with the \c hold request.
* By default, this request will send a new INVITE to the peer with a
* \c sendonly direction for media, but in case you want to set a
* different direction (\c recvonly or \c inactive ) you can do that by
* passing a \c direction attribute as well:
*
\verbatim
{
"request" : "hold",
"direction" : "<sendonly, recvonly or inactive>"
}
\endverbatim
*
* No WebRTC renegotiation will be involved here on the holder side, as
* this will only trigger a re-INVITE on the SIP side. To remove the
* call from on-hold, just send a \c unhold request to the plugin,
* which requires no additional attributes:
*
\verbatim
{
"request" : "unhold"
}
\endverbatim
*
* and will restore the media direction that was set in the SDP before
* putting the call on-hold.
*
* The \c message request allows you to send a SIP MESSAGE to the peer.
* By default, it is sent in dialog, during active call.
* But, if the user is registered, it might be sent out of dialog also. In that case the uri parameter is required.
*
\verbatim
{
"request" : "message",
"call_id" : "<user-defined value of Call-ID SIP header used to send the message; optional>",
"content_type" : "<content type; optional>"
"content" : "<text to send>",
"uri" : "<SIP URI of the peer; optional; if set, the message will be sent out of dialog>",
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP MESSAGE; optional>"
}
\endverbatim
*
* A \c messagesent event will be sent back. Incoming SIP MESSAGEs, instead,
* are notified in \c message events:
*
\verbatim
{
"sip" : "event",
"result" : {
"event" : "message",
"sender" : "<SIP URI of the message sender>",
"displayname" : "<display name of the sender, if available; optional>",
"content_type" : "<content type of the message>",
"content" : "<content of the message>",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* After delivery a \c messagedelivery event will be sent back with the SIP server response.
* Used to track the delivery status of the message.
*
\verbatim
{
"sip" : "event",
"call_id" : "<value of SIP Call-ID header for related message>",
"result" : {
"event" : "messagedelivery",
"code" : "<SIP error code>",
"reason" : "<SIP error reason>",
}
}
\endverbatim
*
* SIP INFO works pretty much the same way, except that you use an \c info
* request to one to the peer:
*
\verbatim
{
"request" : "info",
"type" : "<content type>"
"content" : "<message to send>",
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP INFO; optional>"
}
\endverbatim
*
* A \c infosent event will be sent back. Incoming SIP INFOs, instead,
* are notified in \c info events:
*
\verbatim
{
"sip" : "event",
"result" : {
"event" : "info",
"sender" : "<SIP URI of the message sender>",
"displayname" : "<display name of the sender, if available; optional>",
"type" : "<content type of the message>",
"content" : "<content of the message>",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* As anticipated, SIP events are supported as well, using the SUBSCRIBE
* and NOTIFY mechanism. To do that, you need to use the \c subscribe
* request, which has to be formatted like this:
*
\verbatim
{
"request" : "subscribe",
"call_id" : "<user-defined value of Call-ID SIP header used in all SIP requests throughout the subscription; optional>",
"event" : "<the event to subscribe to, e.g., 'message-summary'; mandatory>",
"accept" : "<what should be put in the Accept header; optional>",
"to" : "<who should be the SUBSCRIBE addressed to; optional, will use the user's identity if missing>",
"subscribe_ttl" : "<integer; number of seconds after which the subscription should expire; optional>",
"headers" : "<array of key/value objects, to specify custom headers to add to the SIP SUBSCRIBE; optional>"
}
\endverbatim
*
* A \c subscribing event will be sent back, followed by a \c subscribe_succeeded if
* the SUBSCRIBE request was accepted, and a \c subscribe_failed if the transaction
* failed instead. Incoming SIP NOTIFY events, instead, are notified in \c notify events:
*
\verbatim
{
"sip" : "event",
"call_id" : "<value of SIP Call-ID header for related subscription>",
"result" : {
"event" : "notify",
"notify" : "<name of the event that the user is subscribed to, e.g., 'message-summary'>",
"substate" : "<substate of the subscription, e.g., 'active'>",
"content-type" : "<content-type of the message>"
"content" : "<content of the message>",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* You can also record a SIP call, and it works pretty much the same the
* VideoCall plugin does. Specifically, you make use of the \c recording
* request to either start or stop a recording, using the following syntax:
*
\verbatim
{
"request" : "recording",
"action" : "<start|stop, depending on whether you want to start or stop recording something>"
"audio" : <true|false; whether or not our audio should be recorded>,
"video" : <true|false; whether or not our video should be recorded>,
"peer_audio" : <true|false; whether or not our peer's audio should be recorded>,
"peer_video" : <true|false; whether or not our peer's video should be recorded>,
"send_peer_pli" : <true|false; whether or not send PLI to request keyframe from peer>,
"filename" : "<base path/filename to use for all the recordings>"
}
\endverbatim
*
* As you can see, this means that the two sides of conversation are recorded
* separately, and so are the audio and video streams if available. You can
* choose which ones to record, in case you're interested in just a subset.
* The \c filename part is just a prefix, and dictates the actual filenames
* that will be used for the up-to-four recordings that may need to be enabled.
*
* A \c recordingupdated event is sent back in case the request is successful.
*
* \section sipmc Simultaneous SIP calls using the same account
*
* As anticipated in the previous sections, attaching to the SIP plugin
* with a Janus handle means creating a SIP stack on behalf of a user
* or application: this typically means registering an account, and being
* able to start or receive calls, handle subscriptions, and so on. This
* also means that, since in Janus each core handle can only be associated
* with a single PeerConnection, each SIP account is limited to a single
* call per time: if a user is in a SIP session already, and another call
* comes in, it's automatically rejected with a \c 486 \c Busy .
*
* While usually not a big deal, there are use cases where it might make
* sense to be able to support multiple concurrent calls, and maybe switch
* from one to the other seamlessly. This is possible in the SIP plugin
* using the so-called \c helper sessions. Specifically, \c helper sessions
* work under the assumption that there's a \c master session that is
* registered normally (the "regular" SIP plugin handle, that is), and
* that these \c helper sessions can simply be associated to that: any time
* another concurrent call is needed, if the \c master session is busy
* one of the \c helpers can be used; the more \c helper sessions are
* available, the more simultaneous calls can be established.
*
* The way this works is simple:
*
* 1. you create a SIP session the usual way, and send a regular \c register
* there; this will be the \c master session, and will return a \c master_id
* when successfully registered;
* 2. for each \c helper you want to add, you attach a new Janus handle
* to the SIP plugin, and send a \c register with \c type: \c "helper" and
* providing the same \c username as the master, plus a \c master_id attribute
* referencing the main session;
* 3. at this point, the new \c helper is associated to the \c master ,
* meaning it can be used to start new calls or receive calls exactly
* as the main session, and using the same account information, credentials,
* etc.
*
* Notice that, as soon as the \c master unregisters, or the Janus handle
* it's on is detached, all the \c helper sessions associated to it are
* automatically torn down as well. Specifically, the plugin will forcibly
* detach the related handles. Should you need to register again, and want
* some helpers there too, you'll have to add them again.
*
* If you want to see this in practice, the SIP plugin demo has a "hidden"
* function you can invoke from the JavaScript console to play with helpers:
* calling the \c addHelper() function will add a new helper, and show additional
* controls. You can add as many helpers as you want.
*
* \section siptr Attended and blind transfers
*
* The Janus SIP plugin supports both attended and blind transfers, and to
* do so mostly relies on the multiple calls functionality: as such, make
* sure you've read and are familiar with the section on \ref sipmc .
*
* Most of the transfer-related functionality are based on existing messages
* and events already documented in the previous section, but there are a
* few aspects you need to be aware of. First of all, if you're the transferor,
* you need to use a new request called \c transfer , that allows you to
* send a SIP REFER to the transferee so to reach a different target. The
* \c transfer request must be formatted like this:
*
\verbatim
{
"request" : "transfer",
"uri" : "<SIP URI to send the transferee too>",
"replace" : "<call-ID of the call this attended transfer is supposed to replace; default is none, which means blind/unattended transfer>"
}
\endverbatim
*
* Whether this is a blind (no call to replace) or attended transfer,
* a \c transferring event will be sent back, as this is an asynchronous
* request. Further updates will come in the form of NOTIFY-related events,
* as a REFER implicitly creates a subscription.
*
* The recipient of a REFER, instead, will receive an asynchronous event
* called \c transfer as well, with info it needs to be aware of. In fact,
* the SIP plugin doesn't do anything automatically: an incoming REFER is
* notified to the application, so that it can decide whether to follow
* up on the transfer or not. The syntax of the event is the following:
*
\verbatim
{
"sip" : "event",
"result" : {
"event" : "transfer",
"refer_id" : <unique ID, internal to Janus, of this referral>,
"refer_to" : "<SIP URI to call>",
"referred_by" : "<SIP URI SIP URI header conveying the identity of the transferor; optional>",
"replaces" : "<call-ID of the call this transfer is supposed to replace; optional, and only present for attended transfers>",
"headers" : "<object with key/value strings; custom headers extracted from SIP event based on incoming_header_prefix defined in register request; optional>"
}
}
\endverbatim
*
* The most important property in that list is \c refer_id as that value
* must be included in the \c call request to call the target, if the
* transfer is accepted: in fact, that's the only way the SIP plugin has
* to correlate the new outgoing call to the previous transfer request,
* and thus be able to notify the transferor about how the call is
* proceeding by means of NOTIFY events. Notice that, if the transferee
* decides to follow up on the transfer request, and they're already in
* a call (e.g., with the transferor), then they must use a different
* handle for the purpose, e.g., via a helper as described in the
* \ref sipmc section.
*
* The transfer target will receive the call exactly as previously discussed,
* with the difference that it may or may not include a \c referred_by
* property for information purposes. Just as the transferee, if they're
* already in a call, it's up to the application to create a helper to
* setup a new Janus handle to accept the transfer.
*
* Notice that the plugin will NOT put the involved calls on-hold, or
* automatically close calls that are meant to be replaced by a transfer.
* All this is the application responsibility, and as such it's up to
* the developer to react to events accordingly.
*
*/
#include "plugin.h"
#include <arpa/inet.h>
#include <net/if.h>
#include <jansson.h>
#include <sofia-sip/msg_header.h>
#include <sofia-sip/nua.h>
#include <sofia-sip/nua_tag.h>
#include <sofia-sip/sdp.h>
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wstrict-prototypes"
#include <sofia-sip/sip_header.h>
#pragma GCC diagnostic pop
#include <sofia-sip/sip_status.h>
#include <sofia-sip/url.h>
#include <sofia-sip/tport_tag.h>
#include <sofia-sip/su_log.h>
#include <sofia-sip/sofia_features.h>
#include "../debug.h"
#include "../apierror.h"
#include "../config.h"
#include "../mutex.h"
#include "../record.h"
#include "../rtp.h"
#include "../rtpsrtp.h"
#include "../rtcp.h"
#include "../sdp-utils.h"
#include "../utils.h"
#include "../ip-utils.h"
/* Plugin information */
#define JANUS_SIP_VERSION 8
#define JANUS_SIP_VERSION_STRING "0.0.8"
#define JANUS_SIP_DESCRIPTION "This is a simple SIP plugin for Janus, allowing WebRTC peers to register at a SIP server and call SIP user agents through a Janus instance."
#define JANUS_SIP_NAME "JANUS SIP plugin"
#define JANUS_SIP_AUTHOR "Meetecho s.r.l."
#define JANUS_SIP_PACKAGE "janus.plugin.sip"
/* Plugin methods */
janus_plugin *create(void);
int janus_sip_init(janus_callbacks *callback, const char *config_path);
void janus_sip_destroy(void);
int janus_sip_get_api_compatibility(void);
int janus_sip_get_version(void);
const char *janus_sip_get_version_string(void);
const char *janus_sip_get_description(void);
const char *janus_sip_get_name(void);
const char *janus_sip_get_author(void);
const char *janus_sip_get_package(void);
void janus_sip_create_session(janus_plugin_session *handle, int *error);
struct janus_plugin_result *janus_sip_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
void janus_sip_setup_media(janus_plugin_session *handle);
void janus_sip_incoming_rtp(janus_plugin_session *handle, janus_plugin_rtp *packet);
void janus_sip_incoming_rtcp(janus_plugin_session *handle, janus_plugin_rtcp *packet);
void janus_sip_hangup_media(janus_plugin_session *handle);
void janus_sip_destroy_session(janus_plugin_session *handle, int *error);
json_t *janus_sip_query_session(janus_plugin_session *handle);
/* Plugin setup */
static janus_plugin janus_sip_plugin =
JANUS_PLUGIN_INIT (
.init = janus_sip_init,
.destroy = janus_sip_destroy,
.get_api_compatibility = janus_sip_get_api_compatibility,
.get_version = janus_sip_get_version,
.get_version_string = janus_sip_get_version_string,
.get_description = janus_sip_get_description,
.get_name = janus_sip_get_name,
.get_author = janus_sip_get_author,
.get_package = janus_sip_get_package,
.create_session = janus_sip_create_session,
.handle_message = janus_sip_handle_message,
.setup_media = janus_sip_setup_media,
.incoming_rtp = janus_sip_incoming_rtp,
.incoming_rtcp = janus_sip_incoming_rtcp,
.hangup_media = janus_sip_hangup_media,
.destroy_session = janus_sip_destroy_session,
.query_session = janus_sip_query_session,
);
/* Plugin creator */
janus_plugin *create(void) {
JANUS_LOG(LOG_VERB, "%s created!\n", JANUS_SIP_NAME);
return &janus_sip_plugin;
}
/* Parameter validation */
static struct janus_json_parameter request_parameters[] = {
{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
};
static struct janus_json_parameter register_parameters[] = {
{"type", JSON_STRING, 0},
{"send_register", JANUS_JSON_BOOL, 0},
{"force_udp", JANUS_JSON_BOOL, 0},
{"force_tcp", JANUS_JSON_BOOL, 0},
{"sips", JANUS_JSON_BOOL, 0},
{"rfc2543_cancel", JANUS_JSON_BOOL, 0},
{"username", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"secret", JSON_STRING, 0},
{"ha1_secret", JSON_STRING, 0},
{"authuser", JSON_STRING, 0},
{"display_name", JSON_STRING, 0},
{"user_agent", JSON_STRING, 0},
{"headers", JSON_OBJECT, 0},
{"contact_params", JSON_OBJECT, 0},
{"master_id", JANUS_JSON_INTEGER, 0},
{"refresh", JANUS_JSON_BOOL, 0},
{"incoming_header_prefixes", JSON_ARRAY, 0},
{"register_ttl", JANUS_JSON_INTEGER, 0}
};
static struct janus_json_parameter subscribe_parameters[] = {
{"to", JSON_STRING, 0},
{"event", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"accept", JSON_STRING, 0},
{"subscribe_ttl", JANUS_JSON_INTEGER, 0},
{"call_id", JANUS_JSON_STRING, 0},
{"headers", JSON_OBJECT, 0}
};
static struct janus_json_parameter proxy_parameters[] = {
{"proxy", JSON_STRING, 0},
{"outbound_proxy", JSON_STRING, 0}
};
static struct janus_json_parameter call_parameters[] = {
{"uri", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"headers", JSON_OBJECT, 0},
{"call_id", JANUS_JSON_STRING, 0},
{"srtp", JSON_STRING, 0},
{"srtp_profile", JSON_STRING, 0},
{"autoaccept_reinvites", JANUS_JSON_BOOL, 0},
{"refer_id", JANUS_JSON_INTEGER, 0},
/* The following are only needed in case "guest" registrations
* still need an authenticated INVITE for some reason */
{"secret", JSON_STRING, 0},
{"ha1_secret", JSON_STRING, 0},
{"authuser", JSON_STRING, 0}
};
static struct janus_json_parameter accept_parameters[] = {
{"srtp", JSON_STRING, 0},
{"headers", JSON_OBJECT, 0},
{"autoaccept_reinvites", JANUS_JSON_BOOL, 0}
};
static struct janus_json_parameter decline_parameters[] = {
{"code", JANUS_JSON_INTEGER, 0},
{"headers", JSON_OBJECT, 0},
{"refer_id", JANUS_JSON_INTEGER, 0}
};
static struct janus_json_parameter transfer_parameters[] = {
{"uri", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"replace", JANUS_JSON_STRING, 0}
};
static struct janus_json_parameter hold_parameters[] = {
{"direction", JSON_STRING, 0}
};
static struct janus_json_parameter recording_parameters[] = {
{"action", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"audio", JANUS_JSON_BOOL, 0},
{"video", JANUS_JSON_BOOL, 0},
{"peer_audio", JANUS_JSON_BOOL, 0},
{"peer_video", JANUS_JSON_BOOL, 0},
{"send_peer_pli", JANUS_JSON_BOOL, 0},
{"filename", JSON_STRING, 0}
};
static struct janus_json_parameter dtmf_info_parameters[] = {
{"digit", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"duration", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
{"headers", JSON_OBJECT, 0}
};
static struct janus_json_parameter info_parameters[] = {
{"type", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"content", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"headers", JSON_OBJECT, 0}
};
static struct janus_json_parameter sipmessage_parameters[] = {
{"content_type", JSON_STRING, 0},
{"content", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
{"uri", JSON_STRING, 0},
{"headers", JSON_OBJECT, 0},
{"call_id", JANUS_JSON_STRING, 0}
};
/* Useful stuff */
static volatile gint initialized = 0, stopping = 0;
static gboolean notify_events = TRUE;
static gboolean ipv6_disabled = FALSE;
static janus_callbacks *gateway = NULL;
static char *local_ip = NULL, *sdp_ip = NULL, *local_media_ip = NULL;
static janus_network_address janus_network_local_media_ip = { 0 };
static int keepalive_interval = 120;
static gboolean behind_nat = FALSE;
static char *user_agent;
#define JANUS_DEFAULT_REGISTER_TTL 3600
static int register_ttl = JANUS_DEFAULT_REGISTER_TTL;
#define JANUS_DEFAULT_SUBSCRIBE_TTL 3600
static int subscribe_ttl = JANUS_DEFAULT_SUBSCRIBE_TTL;
static uint16_t rtp_range_min = 10000;
static uint16_t rtp_range_max = 60000;
static int dscp_audio_rtp = 0;
static int dscp_video_rtp = 0;
static char *sips_certs_dir = NULL;
#define JANUS_DEFAULT_SIP_TIMER_T1X64 32000
static int sip_timer_t1x64 = JANUS_DEFAULT_SIP_TIMER_T1X64;
static gboolean query_contact_header = FALSE;
static GThread *handler_thread;
static void *janus_sip_handler(void *data);
static void janus_sip_hangup_media_internal(janus_plugin_session *handle);
typedef struct janus_sip_message {
janus_plugin_session *handle;
char *transaction;
json_t *message;
json_t *jsep;
} janus_sip_message;
static GAsyncQueue *messages = NULL;
static janus_sip_message exit_message;
typedef enum {
janus_sip_registration_status_disabled = -2,
janus_sip_registration_status_failed = -1,
janus_sip_registration_status_unregistered = 0,
janus_sip_registration_status_registering,
janus_sip_registration_status_registered,
janus_sip_registration_status_unregistering,
} janus_sip_registration_status;
static const char *janus_sip_registration_status_string(janus_sip_registration_status status) {
switch(status) {
case janus_sip_registration_status_disabled:
return "disabled";
case janus_sip_registration_status_failed:
return "failed";
case janus_sip_registration_status_unregistered:
return "unregistered";
case janus_sip_registration_status_registering:
return "registering";
case janus_sip_registration_status_registered:
return "registered";
case janus_sip_registration_status_unregistering:
return "unregistering";
default:
return "unknown";
}
}
typedef enum {
janus_sip_call_status_idle = 0,
janus_sip_call_status_inviting,
janus_sip_call_status_invited,
janus_sip_call_status_incall,
janus_sip_call_status_incall_reinviting,
janus_sip_call_status_incall_reinvited,
janus_sip_call_status_closing,
} janus_sip_call_status;
static const char *janus_sip_call_status_string(janus_sip_call_status status) {
switch(status) {
case janus_sip_call_status_idle:
return "idle";
case janus_sip_call_status_inviting:
return "inviting";
case janus_sip_call_status_invited:
return "invited";
case janus_sip_call_status_incall:
return "incall";
case janus_sip_call_status_incall_reinviting:
return "incall_reinviting";
case janus_sip_call_status_incall_reinvited:
return "incall_reinvited";
case janus_sip_call_status_closing:
return "closing";
default:
return "unknown";
}
}
/* Sofia stuff */
typedef struct ssip_s ssip_t;
typedef struct ssip_oper_s ssip_oper_t;
#undef SU_ROOT_MAGIC_T
#define SU_ROOT_MAGIC_T ssip_t
#undef NUA_MAGIC_T
#define NUA_MAGIC_T ssip_t
#undef NUA_HMAGIC_T
#define NUA_HMAGIC_T ssip_oper_t
struct ssip_s {
su_home_t s_home[1];
su_root_t *s_root;
nua_t *s_nua;
nua_handle_t *s_nh_r, *s_nh_i, *s_nh_m;
char *contact_header; /* Only needed for Sofia SIP >= 1.13 */
GHashTable *subscriptions;
janus_mutex smutex;
struct janus_sip_session *session;
};
typedef struct janus_sip_transfer {
struct janus_sip_session *session;
char *referred_by;
char *custom_headers;
nua_handle_t *nh_s;
nua_saved_event_t saved[1];
} janus_sip_transfer;
typedef enum {
janus_sip_secret_type_plaintext = 1,
janus_sip_secret_type_hashed = 2,
janus_sip_secret_type_unknown
} janus_sip_secret_type;
typedef struct janus_sip_account {
char *identity;
char *user_agent; /* Used to override the general UA string */
gboolean force_udp;
gboolean force_tcp;
gboolean sips;
gboolean rfc2543_cancel;
char *username;
char *display_name; /* Used for outgoing calls in the From header */
char *authuser; /**< username to use for authentication */
char *secret;
janus_sip_secret_type secret_type;
int sip_port;
char *proxy;
char *outbound_proxy;