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Tijmen de Mes, 04/19/2012 05:05 PM
Middleware API¶
This chapter describes the Middleware API for SIP SIMPLE client SDK that can be used for developing a user interface (e.g. Graphical User Interface). The Middleware provides a non-blocking API that communicates with the user interface asynchronously by using Notifications. For its configuration, the Middleware uses the Configuration API.
SIPApplication¶
Implemented in [browser:sipsimple/application.py]
Implements a high-level application responsable for starting and stopping various sub-systems required to implement a fully featured SIP User Agent application. The SIPApplication class is a Singleton and can be instantiated from any part of the code, obtaining a reference to the same object. The SIPApplication takes care of initializing the following components:- the twisted thread
- the configuration system, via the ConfigurationManager
- the core Engine using the settings in the configuration
- the AccountManager, using the accounts in the configuration
- the SessionManager, in order to handle incoming sessions
- two AudioBridges, using the settings in the configuration
The attributes in this class can be set and accessed on both this class and its subclasses, as they are implemented using descriptors which keep single value for each attribute, irrespective of the class from which that attribute is set/accessed. Usually, all attributes should be considered read-only.
methods¶
__init__(self)
Instantiates a new SIPApplication.
start(self, storage)
Starts the
SIPApplication
which initializes all the components in the correct order. Thestorage
is saved as an attribute which other entities like theConfiguration Manager
will use to take the appropriate backend. If any error occurs with loading the configuration, the exception raised by theConfigurationManager
is propagated by this method andSIPApplication
can be started again. After this, any fatal errors will result in the SIPApplication being stopped and unusable, which means the whole application will need to stop. This method returns as soon as the twisted thread has been started, which means the application must wait for theSIPApplicationDidStart
notification in order to know that the application started.
stop(self)
Stop all the components started by the SIPApplication. This method returns immediately, but a
SIPApplicationDidEnd
notification is sent when all the components have been stopped.
attributes¶
running
True
if the SIPApplication is running (it has been started and it has not been told to stop),False
otherwise.
storage
Holds an object which implements the
ISIPSimpleStorage
interface which will be used to provide a storage facility to other middleware components.
local_nat_type
String containing the detected local NAT type.
alert_audio_mixer
The
AudioMixer
object created on the alert audio device as defined by the configuration (by SIPSimpleSettings.audio.alert_device).
alert_audio_bridge
An
AudioBridge
whereIAudioPort
objects can be added to playback sound to the alert device.
alert_audio_device
An
AudioDevice
which corresponds to the alert device as defined by the configuration. This will always be part of the alert_audio_bridge.
voice_audio_mixer
The
AudioMixer
object created on the voice audio device as defined by the configuration (by SIPSimpleSettings.audio.input_device and SIPSimpleSettings.audio.output_device).
voice_audio_bridge
An
AudioBridge
whereIAudioPort
objects can be added to playback sound to the output device or record sound from the input device.
voice_audio_device
An
AudioDevice
which corresponds to the voice device as defined by the configuration. This will always be part of the voice_audio_bridge.
notifications¶
SIPApplicationWillStart
This notification is sent just after the configuration has been loaded and the twisted thread started, but before any other components have been initialized.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPApplicationDidStart
This notification is sent when all the components have been initialized. Note: it doesn't mean that all components have succeeded, for example, the account might not have registered by this time, but the registration process will have started.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPApplicationWillEnd
This notification is sent as soon as the
stop()
method has been called.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPApplicationDidEnd
This notification is sent when all the components have been stopped. All components have been given reasonable time to shutdown gracefully, such as the account unregistering. However, because of factors outside the control of the middleware, such as network problems, some components might not have actually shutdown gracefully; this is needed because otherwise the SIPApplication could hang indefinitely (for example because the system is no longer connected to a network and it cannot be determined when it will be again).
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPApplicationFailedToStartTLS
This notification is sent when a problem arises with initializing the TLS transport. In this case, the Engine will be started without TLS support and this notification contains the error which identifies the cause for not being able to start the TLS transport.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
error:
The exception raised by the Engine which identifies the cause for not being able to start the TLS transport.
Storage API¶
Different middleware components may need to store data, i.e. configuration files or XCAP documents. The Storage API
defines a collection of backends which other components will use to store their data.
API Definition¶
The Storage API
currently requires the following attributes to be defined as per the ISIPSimpleStorage
interface:
configuration_backend
The backend used for storing the configuration.
xcap_storage_factory
Factory used to create XCAP storage backends for each account.
Provided implementations¶
Two storage implementations are provided: FileStorage and MemoryStorage both located in the sipsimple.storage module.
SIP Sessions¶
SIP sessions are supported by the sipsimple.session.Session
class and independent stream classes, which need to implement the sipsimple.streams.IMediaStream
interface. The Session
class takes care of the signalling, while the streams offer the actual media support which is negotiated by the Session
. The streams which are implemented in the SIP SIMPLE middleware are provided in modules within the sipsimple.streams
package, but they are accessible for import directly from sipsimple.streams
. Currently, the middleware implements two types of streams, one for RTP data, with a concrete implementation in the AudioStream
class, and one for MSRP sessions, with concrete implementations in the ChatStream
, FileTransferStream
and DesktopSharingStream
classes. However, the application can provide its own stream implementation, provided they respect the IMediaStream
interface.
The sipsimple.streams
module also provides a mechanism for automatically registering media streams in order for them to be used for incoming sessions. This is explained in more detail in MediaStreamRegistry.
SessionManager¶
Implemented in [browser:sipsimple/session.py]
The sipsimple.session.SessionManager
class is a singleton, which acts as the central aggregation point for sessions within the middleware.
Although it is mainly used internally, the application can use it to query information about all active sessions.
The SessionManager is implemented as a singleton, meaning that only one instance of this class exists within the middleware. The SessionManager is started by the SIPApplication and takes care of handling incoming sessions and closing all sessions when SIPApplication is stopped.
attributes¶
sessions
A property providing a copy of the list of all active
Sesssion
objects within the application, meaning anySession
object that exists globally within the application and is not in theNULL
orTERMINATED
state.
methods¶
__init__(self)
Instantiate a new
SessionManager
object.
start(self)
Start the
SessionManager
in order to be able to handle incoming sessions. This method is called automatically when SIPApplication is started. The application should not call this method directly.
stop(self)
End all connected sessions. This method is called automatically when SIPApplication is stopped. The application should not call this method directly.
Session¶
Implemented in [browser:sipsimple/session.py]
A sipsimple.session.Session
object represents a complete SIP session between the local and a remote endpoints. Both incoming and outgoing sessions are represented by this class.
A Session
instance is a stateful object, meaning that it has a state
attribute and that the lifetime of the session traverses different states, from session creation to termination. State changes are triggered by methods called on the object by the application or by received network events. These states and their transitions are represented in the following diagram:
Although these states are crucial to the correct operation of the Session
object, an application using this object does not need to keep track of these states, as a set of notifications is also emitted, which provide all the necessary information to the application.
The Session
is completely independent of the streams it contains, which need to be implementations of the sipsimple.streams.IMediaStream
interface. This interface provides the API by which the Session
communicates with the streams. This API should not be used by the application, unless it also provides stream implementations or a SIP INVITE session implementation.
methods¶
__init__(self, account)
Creates a new
Session
object in theNone
state.
account:
The local account to be associated with this
Session
.
connect(self, to_header, routes, streams, is_focus=False
, subject=None
)
Will set up the
Session
as outbound and propose the new session to the specified remote party and move the state machine to theoutgoing
state.
Before contacting the remote party, aSIPSessionNewOutgoing
notification will be emitted.
If there is a failure or the remote party rejected the offer, aSIPSessionDidFail
notification will be sent.
Any time a ringing indication is received from the remote party, aSIPSessionGotRingIndication
notification is sent.
If the remote party accepted the session, aSIPSessionWillStart
notification will be sent, followed by aSIPSessionDidStart
notification when the session is actually established.
This method may only be called while in theNone
state.
to_header:
A
sipsimple.core.ToHeader
object representing the remote identity to initiate the session to.
routes:
An iterable of
sipsimple.util.Route
objects, specifying the IP, port and transport to the outbound proxy.
These routes will be tried in order, until one of them succeeds.
streams:
A list of stream objects which will be offered to the remote endpoint.
is_focus:
Boolean flag indicating if the
isfocus
parameter should be added to theContact
header according to RFC 4579.
subject:
Session subject. If not None a
Subject
header will be added with the specified value.
send_ring_indication(self)
Sends a 180 provisional response in the case of an incoming session.
accept(self, streams)
Calling this methods will accept an incoming session and move the state machine to the
accepting
state.
When there is a new incoming session, aSIPSessionNewIncoming
notification is sent, after which the application can call this method on the sender of the notification.
After this method is called,SIPSessionWillStart
followed bySIPSessionDidStart
will be emitted, orSIPSessionDidFail
on an error.
This method may only be called while in theincoming
state.
streams:
A list of streams which needs to be a subset of the proposed streams which indicates which streams are to be accepted. All the other proposed streams will be rejected.
reject(self, code=603
, reason=None
)
Reject an incoming session and move it to the
terminating
state, which eventually leads to theterminated
state.
Calling this method will cause theSession
object to emit aSIPSessionDidFail
notification once the session has been rejected.
This method may only be called while in theincoming
state.
code:
An integer which represents the SIP status code in the response which is to be sent. Usually, this is either 486 (Busy) or 603 (Decline/Busy Everywhere).
reason:
The string which is to be sent as the SIP status reason in the response, or None if PJSIP's default reason for the specified code is to be sent.
accept_proposal(self, streams)
When the remote party proposes to add some new streams, signaled by the
SIPSessionGotProposal
notification, the application can use this method to accept the stream(s) being proposed.
After calling this method aSIPSessionGotAcceptProposal
notification is sent, unless an error occurs while setting up the new stream, in which case aSIPSessionHadProposalFailure
notification is sent and a rejection is sent to the remote party. As with any action which causes the streams in the session to change, aSIPSessionDidRenegotiateStreams
notification is also sent.
This method may only be called while in thereceived_proposal
state.
streams:
A list of streams which needs to be a subset of the proposed streams which indicates which streams are to be accepted. All the other proposed streams will be rejected.
reject_proposal(self, code=488
, reason=None
)
When the remote party proposes new streams that the application does not want to accept, this method can be used to reject the proposal, after which a
SIPSessionGotRejectProposal
orSIPSessionHadProposalFailure
notification is sent.
This method may only be called while in thereceived_proposal
state.
code:
An integer which represents the SIP status code in the response which is to be sent. Usually, this is 488 (Not Acceptable Here).
reason:
The string which is to be sent as the SIP status reason in the response, or None if PJSIP's default reason for the specified code is to be sent.
add_stream(self, stream)
Proposes a new stream to the remote party.
Calling this method will cause aSIPSessionGotProposal
notification to be emitted.
After this, the state machine will move into thesending_proposal
state until either aSIPSessionGotAcceptProposal
,SIPSessionGotRejectProposal
orSIPSessionHadProposalFailure
notification is sent, informing the application if the remote party accepted the proposal. As with any action which causes the streams in the session to change, aSIPSessionDidRenegotiateStreams
notification is also sent.
This method may only be called while in theconnected
state.
remove_stream(self, stream)
Stop the stream and remove it from the session, informing the remote party of this. Although technically this is also done via an SDP negotiation which may fail, the stream will always get remove (if the remote party refuses the re-INVITE, the result will be that the remote party will have a different view of the active streams than the local party).
This method may only be called while in theconnected
state.
cancel_proposal(self)
This method cancels a proposal of adding a stream to the session by sending a CANCEL request. A
SIPSessionGotRejectProposal
notification will be sent with code 487.
hold(self)
Put the streams of the session which support the notion of hold on hold.
This will cause aSIPSessionDidChangeHoldState
notification to be sent.
This method may be called in any state and will send the re-INVITE as soon as it is possible.
unhold(self)
Take the streams of the session which support the notion of hold out of hold.
This will cause aSIPSessionDidChangeHoldState
notification to be sent.
This method may be called in any state and will send teh re-INVITE as soon as it is possible.
end(self)
This method may be called any time after the
Session
has started in order to terminate the session by sending a BYE request.
Right before termination aSIPSessionWillEnd
notification is sent, after terminationSIPSessionDidEnd
is sent.
transfer(self, target_uri, replaced_session=None
)
Proposes a blind call transfer to a new target URI or assisted transfer to an URI belonging to an already established session.
accept_transfer(self)
Accepts an incoming call transfer request.
reject_transfer(self, code=486
, *reason_=None
)
Rejects an incoming call transfer request.
attributes¶
state
The state the object is currently in, being one of the states from the diagram above.
account
The
sipsimple.account.Account
orsipsimple.account.BonjourAccount
object that theSession
is associated with.
On an outbound session, this is the account the application specified on object instantiation.
direction
A string indicating the direction of the initial negotiation of the session.
This can be eitherNone
, "incoming" or "outgoing".
transport
A string representing the transport this
Session
is using:"udp"
,"tcp"
or"tls"
.
start_time
The time the session started as a
datetime.datetime
object, orNone
if the session was not yet started.
stop_time
The time the session stopped as a
datetime.datetime
object, orNone
if the session has not yet terminated.
on_hold
Boolean indicating whether the session was put on hold, either by the local or the remote party.
remote_user_agent
A string indicating the remote user agent, if it provided one.
Initially this will beNone
, it will be set as soon as this information is received from the remote party (which may be never).
local_identity
The
sipsimple.core.FrozenFromHeader
orsipsimple.core.FrozenToHeader
identifying the local party, if the session is active,None
otherwise.
remote_identity
The
sipsimple.core.FrozenFromHeader
orsipsimple.core.FrozenToHeader
identifying the remote party, if the session is active,None
otherwise.
streams
A list of the currently active streams in the
Session
.
proposed_streams
A list of the currently proposed streams in the
Session
, orNone
if there is no proposal in progress.
conference
A
ConferenceHandler
object instance (or Null). It can be later used to add/remove participants from a remote conference.
subject
The session subject as a unicode object.
replaced_session
A
Session
object instance (or Null). It can be used for assisted call transfer.
transfer_handler
A
TransferHandler
object instance (or Null). It is used for managing the call transfer process.
transfer_info
A
TransferInfo
object instance (or Null). It is used for describing the details of a call transfer operation.
notifications¶
SIPSessionNewIncoming
Will be sent when a new incoming
Session
is received.
The application should listen for this notification to get informed of incoming sessions.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
streams:
A list of streams that were proposed by the remote party.
SIPSessionNewOutgoing
Will be sent when the application requests a new outgoing
Session
.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
streams:
A list of streams that were proposed to the remote party.
SIPSessionGotRingIndication
Will be sent when an outgoing
Session
receives an indication that a remote device is ringing.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPSessionGotProvisionalResponse
Will be sent whenever the
Session
receives a provisional response as a result of sending a (re-)INVITE.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
code:
The SIP status code received.
reason:
The SIP status reason received.
SIPSessionWillStart
Will be sent just before a
Session
completes negotiation.
In terms of SIP, this is sent after the final response to theINVITE
, but before theACK
.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPSessionDidStart
Will be sent when a
Session
completes negotiation and all the streams have started.
In terms of SIP this is sent after theACK
was sent or received.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
streams:
The list of streams which now form the active streams of the
Session
.
SIPSessionDidFail
This notification is sent whenever the session fails before it starts.
The failure reason is included in the data attributes.
This notification is never followed bySIPSessionDidEnd
.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
originator:
A string indicating the originator of the
Session
. This will either be "local" or "remote".
code:
The SIP error code of the failure.
reason:
A SIP status reason.
failure_reason:
A string which represents the reason for the failure, such as
"user_request"
,"missing ACK"
,"SIP core error..."
.
SIPSessionWillEnd
Will be sent just before terminating a
Session
.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPSessionDidEnd
Will be sent always when a
Session
ends as a result of remote or local session termination.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
originator:
A string indicating who originated the termination. This will either be "local" or "remote".
end_reason:
A string representing the termination reason, such as
"user_request"
,"SIP core error..."
.
SIPSessionDidChangeHoldState
Will be sent when the session got put on hold or removed from hold, either by the local or the remote party.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
originator:
A string indicating who originated the hold request, and consequently in which direction the session got put on hold.
on_hold:
True
if there is at least one stream which is on hold andFalse
otherwise.
partial:
True
if there is at least one stream which is on hold and one stream which supports hold but is not on hold andFalse
otherwise.
SIPSessionGotProposal
Will be sent when either the local or the remote party proposes to add streams to the session.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
originator:
The party that initiated the stream proposal, can be either "local" or "remote".
streams:
A list of streams that were proposed.
SIPSessionGotRejectProposal
Will be sent when either the local or the remote party rejects a proposal to have streams added to the session.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
originator:
The party that initiated the stream proposal, can be either "local" or "remote".
code:
The code with which the proposal was rejected.
reason:
The reason for rejecting the stream proposal.
streams:
The list of streams which were rejected.
SIPSessionGotAcceptProposal
Will be sent when either the local or the remote party accepts a proposal to have stream( added to the session.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
originator:
The party that initiated the stream proposal, can be either "local" or "remote".
streams:
The list of streams which were accepted.
proposed_streams:
The list of streams which were originally proposed.
SIPSessionHadProposalFailure
Will be sent when a re-INVITE fails because of an internal reason (such as a stream not being able to start).
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
failure_reason:
The error which caused the proposal to fail.
streams:
The streams which were part of this proposal.
SIPSessionDidRenegotiateStreams
Will be sent when a media stream is either activated or deactivated.
An application should listen to this notification in order to know when a media stream can be used.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
action:
A string which is either
"add"
or"remove"
which specifies what happened to the streams the notificaton referes to
streams:
A list with the streams which were added or removed.
SIPSessionDidProcessTransaction
Will be sent whenever a SIP transaction is complete in order to provide low-level details of the progress of the INVITE dialog.
timestamp:
Adatetime.datetime
object indicating when the notification was sent.
originator:
The initiator of the transaction,
"local"
or"remote"
.
method:
The method of the request.
code:
The SIP status code of the response.
reason:
The SIP status reason of the response.
ack_received:
This attribute is only present for INVITE transactions and has one of the values
True
,False
or"unknown"
. The last value may occur then PJSIP does not let us know whether the ACK was received or not.
SIPSessionTransferNewOutgoing
Will be sent whenever a SIP session initiates an outgoing call transfer request.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
transfer_destination:
The destination SIP URI of the call transfer request.
transfer_source:
The source SIP URI of the call transfer request.
SIPSessionTransferDidStart
Will be sent whenever a call transfer has been started.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
SIPSessionTransferDidFail
Will be sent whenever a call transfer request has failed.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
code:
The SIP failure code reported by the SIP stack.
reason:
The reason of the failure as a string.
As an example for how to use the Session
object, the following provides a basic Python program that initiates an outgoing SIP session request see Minimalist Session Example code.
IMediaStream¶
Implemented in [browser:sipsimple/streams/+init+.py]
This interface describes the API which the Session
uses to communicate with the streams. All streams used by the Session
must respect this interface.
methods¶
__init__(self, account)
Initializes the generic stream instance.
new_from_sdp(cls, account, remote_sdp, stream_index)
A classmethod which returns an instance of this stream implementation if the sdp is accepted by the stream or None otherwise.
account:
The
sipsimple.account.Account
orsipsimple.account.BonjourAccount
object the session which this stream would be part of is associated with.
remote_sdp:
The
FrozenSDPSession
which was received by the remote offer.
stream_index:
An integer representing the index within the list of media streams within the whole SDP which this stream would be instantiated for.
get_local_media(self, for_offer)
Return an
SDPMediaStream
which represents an offer for using this stream iffor_offer
isTrue
and a response to an SDP proposal otherwise.
for_offer:
True
if theSDPMediaStream
will be used for an SDP proposal andFalse
if for a response.
initialize(self, session, direction)
Initializes the stream. This method will get called as soon as the stream is known to be at least offered as part of the
Session
. If initialization goes fine, the stream must send aMediaStreamDidInitialize
notification or aMediaStreamDidFail
notification otherwise.
session:
The
Session
object this stream will be part of.
direction:
"incoming"
if the stream was created because of a received proposal and"outgoing"
if a proposal was sent. Note that this need not be the same as the initial direction of theSession
since streams can be proposed in either way using re-INVITEs.
start(self, local_sdp, remote_sdp, stream_index)
Starts the stream. This method will be called as soon is known to be used in the
Session
(eg. only called for an incoming proposal if the local party accepts the proposed stream). If starting succeeds, the stream must send aMediaStreamDidStart
notification or aMediaStreamDidFail
notification otherwise.
local_sdp:
The
FrozenSDPSession
which is used by the local endpoint.
remote_sdp:
The
FrozenSDPSession
which is used by the remote endpoint.
stream_index:
An integer representing the index within the list of media streams within the whole SDP which this stream is represented by.
validate_update(self, remote_sdp, stream_index)
This method will be called when a re-INVITE is received which changes the parameters of the stream within the SDP. The stream must return
True
if the changes are acceptable orFalse
otherwise. If any changed streams returnFalse
for a re-INVITE, the re-INVITE will be refused with a negative response. This means that streams must not changed any internal data when this method is called as the update is not guaranteed to be applied even if the stream returnsTrue
.
remote_sdp:
The
FrozenSDPSession
which is used by the remote endpoint.
stream_index:
An integer representing the index within the list of media streams within the whole SDP which this stream is represented by.
update(self, local_sdp, remote_sdp, stream_index)
This method is called when the an SDP negotiation initiated by either the local party or the remote party succeeds. The stream must update its internal state according to the new SDP in use.
local_sdp:
The
FrozenSDPSession
which is used by the local endpoint.
remote_sdp:
The
FrozenSDPSession
which is used by the remote endpoint.
stream_index:
An integer representing the index within the list of media streams within the whole SDP which this stream is represented by.
hold(self)
Puts the stream on hold if supported by the stream. Typically used by audio and video streams. The stream must immediately stop sending/receiving data and calls to
get_local_media()
following calls to this method must return an SDP which reflects the new hold state.
unhold(self)
Takes the stream off hold. Typically used by audio and video streams. Calls to
get_local_media()
following calls to this method must return an SDP which reflects the new hold state.
deactivate(self)
This method is called on a stream just before the stream will be removed from the
Session
(either as a result of a re-INVITE or a BYE). This method is needed because it avoids a race condition with streams using stateful protocols such as TCP: the stream connection might be terminated before the SIP signalling announces this due to network routing inconsistencies and the other endpoint would not be able to distinguish between this case and an error which caused the stream transport to fail. The stream must not take any action, but must consider that the transport being closed by the other endpoint after this method was called as a normal situation rather than an error condition.
end(self)
Ends the stream. This must close the underlying transport connection. The stream must send a
MediaStreamWillEnd
just after this method is called and aMediaStreamDidEnd
as soon as the operation is complete. This method is always be called by theSession
on the stream if at least theinitialize()
method has been called. This means that once a stream sends theMediaStreamDidFail
notification, theSession
will still call this method.
attributes¶
type (class attribute)
A string identifying the stream type (eg:
"audio"
,"video"
).
priority (class attribute)
An integer value indicating the stream priority relative to the other streams types (higher numbers have higher priority).
hold_supported
True if the stream supports hold
on_hold_by_local
True if the stream is on hold by the local party
on_hold_by_remote
True if the stream is on hold by the remote
on_hold
True if either on_hold_by_local or on_hold_by_remote is true
notifications¶
These notifications must be generated by all streams in order for the Session
to know the state of the stream.
MediaStreamDidInitialize
Sent when the stream has been successfully initialized.
MediaStreamDidStart
Sent when the stream has been successfully started.
MediaStreamDidFail
Sent when the stream has failed either as a result of calling one of its methods, or during the normal operation of the stream (such as the transport connection being closed).
MediaStreamWillEnd
Sent immediately after the
end()
method is called.
MediaStreamDidEnd
Sent when the
end()
method finished closing the stream.
MediaStreamRegistrar¶
This is a convenience metaclass which automatically registers a defined class with the MediaStreamRegistry
. In order to use this class, one simply needs to use it as the metaclass of the new stream.
from zope.interface import implements from sipsimple.streams import IMediaStream, MediaStreamRegistrar class MyStream(object): __metaclass__ = MediaStreamRegistrar implements(IMediaStream) [...]
AudioStream¶
Implemented in [browser:sipsimple/streams/rtp.py]
The AudioStream
is an implementation of IMediaStream
which supports audio data using the AudioTransport
and RTPTransport
of the SIP core. As such, it provides all features of these objects, including ICE negotiation. An example SDP created using the AudioStream
is provided below:
Content-Type: application/sdp Content-Length: 1093 v=0 o=- 3467525278 3467525278 IN IP4 192.168.1.6 s=blink-0.10.7-beta c=IN IP4 80.101.96.20 t=0 0 m=audio 55328 RTP/AVP 104 103 102 3 9 0 8 101 a=rtcp:55329 IN IP4 80.101.96.20 a=rtpmap:104 speex/32000 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:esI6DbLY1+Aceu0JNswN9Z10DcFx5cZwqJcu91jb a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:SHuEMm1BYJqOF4udKl73EaCwnsI57pO86bYKsg70 a=ice-ufrag:2701ed80 a=ice-pwd:6f8f8281 a=candidate:S 1 UDP 31 80.101.96.20 55328 typ srflx raddr 192.168.1.6 rport 55328 a=candidate:H 1 UDP 23 192.168.1.6 55328 typ host a=candidate:H 1 UDP 23 10.211.55.2 55328 typ host a=candidate:H 1 UDP 23 10.37.129.2 55328 typ host a=candidate:S 2 UDP 30 80.101.96.20 55329 typ srflx raddr 192.168.1.6 rport 55329 a=candidate:H 2 UDP 22 192.168.1.6 55329 typ host a=candidate:H 2 UDP 22 10.211.55.2 55329 typ host a=candidate:H 2 UDP 22 10.37.129.2 55329 typ host a=sendrecv
As an implementation of IAudioPort
, an AudioStream
can be added to an AudioBridge
to send or to read audio data to/from other audio objects. It is connected to the voice AudioMixer
(SIPApplication.voice_audio_mixer
) so it can only be added to bridges using the same AudioMixer
. It also contains an AudioBridge
on the bridge
attribute which always contains an AudioDevice
corresponding to the input and output devices; when the stream is active (started and not on hold), the bridge also contains the stream itself and when recording is active, the stream contains a WaveRecorder
which records audio data.
methods¶
start_recording(self, filename=None
)
If an audio stream is present within this session, calling this method will record the audio to a
.wav
file.
Note that when the session is on hold, nothing will be recorded to the file.
Right before starting the recording aSIPSessionWillStartRecordingAudio
notification will be emitted, followed by aSIPSessionDidStartRecordingAudio
.
This method may only be called while the stream is started.
filename:
The name of the
.wav
file to record to.
If this is set toNone
, a default file name including the session participants and the timestamp will be generated using the directory defined in the configuration.
stop_recording(self)
This will stop a previously started recording.
Before stopping, aSIPSessionWillStopRecordingAudio
notification will be sent, followed by aSIPSessionDidStopRecordingAudio
.
send_dtmf(self, digit)
If the audio stream is started, sends a DTMF digit to the remote party.
digit:
This should a string of length 1, containing a valid DTMF digit value (0-9, A-D, * or #).
attributes¶
sample_rate
If the audio stream was started, this attribute contains the sample rate of the audio negotiated.
codec
If the audio stream was started, this attribute contains the name of the audio codec that was negotiated.
srtp_active
If the audio stream was started, this boolean attribute indicates if SRTP is currently being used on the stream.
ice_active
True
if the ICE candidates negotiated are being used,False
otherwise.
local_rtp_address
If an audio stream is present within the session, this attribute contains the local IP address used for the audio stream.
local_rtp_port
If an audio stream is present within the session, this attribute contains the local UDP port used for the audio stream.
remote_rtp_address_sdp
If the audio stream was started, this attribute contains the IP address that the remote party gave to send audio to.
remote_rtp_port_sdp
If the audio stream was started, this attribute contains the UDP port that the remote party gave to send audio to.
remote_rtp_address_received
If the audio stream was started, this attribute contains the remote IP address from which the audio stream is being received.
remote_rtp_port_received
If the audio stream was started, this attribute contains the remote UDP port from which the audio stream is being received.
local_rtp_candidate_type
The local ICE candidate type which was selected by the ICE negotiation if it succeeded and
None
otherwise.
remote_rtp_candidate_type
The remote ICE candidate type which was selected by the ICE negotiation if it succeeded and
None
otherwise.
recording_filename
If the audio stream is currently being recorded to disk, this property contains the name of the
.wav
file being recorded to.
notifications¶
AudioStreamDidChangeHoldState
Will be sent when the hold state is changed as a result of either a SIP message received on the network or the application calling the
hold()/unhold()
methods on theSession
this stream is part of.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
originator:
A string representing the party which requested the hold change,
"local"
or"remote"
on_hold:
A boolean indicating the new hold state from the point of view of the originator.
*AudioStreamWillStartRecordingAudio_
Will be sent when the application requested that the audio stream be recorded to a
.wav
file, just before recording starts.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
filename:
The full path to the
.wav
file being recorded to.
AudioStreamDidStartRecordingAudio
Will be sent when the application requested that the audio stream be recorded to a
.wav
file, just after recording started.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
filename:
The full path to the
.wav
file being recorded to.
AudioStreamWillStopRecordingAudio
Will be sent when the application requested ending the recording to a
.wav
file, just before recording stops.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
filename:
The full path to the
.wav
file being recorded to.
AudioStreamDidStopRecordingAudio
Will be sent when the application requested ending the recording to a
.wav
file, just after recording stoped.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
filename:
The full path to the
.wav
file being recorded to.
AudioStreamDidChangeRTPParameters
This notification is sent when the RTP parameters are changed, such as codec, sample rate, RTP port etc.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
AudioStreamGotDTMF
Will be send if there is a DMTF digit received from the remote party on the audio stream.
timestamp:
A
datetime.datetime
object indicating when the notification was sent.
digit:
The DTMF digit that was received, in the form of a string of length 1.
AudioStreamICENegotiationStateDidChange
This notification is proxied from the
RTPTransport
and as such has the same data as theRTPTransportICENegotiationStateDidChange
.
AudioStreamICENegotiationDidSucceed
This notification is proxied from the
RTPTransport
and as such has the same data as theRTPTransportICENegotiationDidSucceed
.
AudioStreamICENegotiationDidFail
This notification is proxied from the
RTPTransport
and as such has the same data as theRTPTransportICENegotiationDidFail
.
AudioStreamDidTimeout
This notification is proxied from the
RTPTransport
. It's sent when the RTP transport did not receive any data after the specified amount of time (rtp.timeout setting in theAccount
).
Updated by Tijmen de Mes over 12 years ago · 168 revisions