[asterisk-scf-commits] asterisk-scf/release/sip.git branch "master" updated.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Tue Aug 2 11:35:13 CDT 2011
branch "master" has been updated
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Summary of changes:
config/SipConfigurator.py | 15 ++
.../SipSessionManager/SipConfigurationIf.ice | 18 ++
src/CMakeLists.txt | 4 +
src/PJSipSessionModule.cpp | 249 +++++++++++++++++++-
src/PJSipSessionModule.h | 4 +
src/SipConfiguration.cpp | 8 +
src/SipEndpoint.cpp | 11 +-
src/SipEndpoint.h | 5 +
src/SipSession.cpp | 138 ++++++++++-
src/SipSession.h | 43 +++-
src/SipTelephonyEventSink.cpp | 220 +++++++++++++++++
src/SipTelephonyEventSink.h | 61 +++++
src/SipTelephonyEventSource.cpp | 108 +++++++++
src/SipTelephonyEventSource.h | 63 +++++
14 files changed, 924 insertions(+), 23 deletions(-)
create mode 100644 src/SipTelephonyEventSink.cpp
create mode 100644 src/SipTelephonyEventSink.h
create mode 100644 src/SipTelephonyEventSource.cpp
create mode 100644 src/SipTelephonyEventSource.h
- Log -----------------------------------------------------------------
commit f738f0c7b03d342bae114b88c46fc0a70d22e5ec
Merge: 753d3c7 9404329
Author: Mark Michelson <mmichelson at digium.com>
Date: Tue Aug 2 11:34:19 2011 -0500
Merge branch 'telephone-events'
Conflicts:
slice/AsteriskSCF/Configuration/SipSessionManager/SipConfigurationIf.ice
src/PJSipSessionModule.cpp
src/PJSipSessionModule.h
src/SipConfiguration.cpp
src/SipEndpoint.cpp
src/SipEndpoint.h
src/SipSession.cpp
src/SipSession.h
diff --cc slice/AsteriskSCF/Configuration/SipSessionManager/SipConfigurationIf.ice
index 715e039,e2a477f..9cdafe8
--- a/slice/AsteriskSCF/Configuration/SipSessionManager/SipConfigurationIf.ice
+++ b/slice/AsteriskSCF/Configuration/SipSessionManager/SipConfigurationIf.ice
@@@ -32,458 -32,426 +32,476 @@@ module SipSessionManage
["suppress"]
module V1
{
- /**
- * Service locator category for finding the configuration service
- */
- const string ConfigurationDiscoveryCategory = "SipConfiguration";
- /**
- * Service locator parameters class for discovering the configuration service
- */
- unsliceable class SipConfigurationParams extends AsteriskSCF::Core::Discovery::V1::ServiceLocatorParams
- {
- /**
- * Unique name for the configuration service
- */
- string name;
- };
+/**
+ * Service locator category for finding the configuration service
+ */
+const string ConfigurationDiscoveryCategory = "SipConfiguration";
- /**
- * Local visitor class for visiting SIP configuration groups
- */
- local class SipConfigurationGroupVisitor extends AsteriskSCF::System::Configuration::V1::ConfigurationGroupVisitor
- {
- };
+/**
+ * Service locator parameters class for discovering the configuration service
+ */
+unsliceable class SipConfigurationParams extends AsteriskSCF::Core::Discovery::V1::ServiceLocatorParams
+{
+ /**
+ * Unique name for the configuration service
+ */
+ string name;
+};
- /**
- * Generic SIP configuration group
- */
- ["visitor:SipConfigurationGroupVisitor"] class SipConfigurationGroup extends AsteriskSCF::System::Configuration::V1::ConfigurationGroup
- {
- };
+/**
+ * Local visitor class for visiting SIP configuration groups
+ */
+local class SipConfigurationGroupVisitor extends AsteriskSCF::System::Configuration::V1::ConfigurationGroupVisitor
+{
+};
- /**
- * General SIP configuration group that contains general items related to the SIP component as a whole
- */
- class SipGeneralGroup extends SipConfigurationGroup
- {
- };
+/**
+ * Generic SIP configuration group
+ */
+["visitor:SipConfigurationGroupVisitor"] class SipConfigurationGroup extends AsteriskSCF::System::Configuration::V1::ConfigurationGroup
+{
+};
- /**
- * Local visitor class for visiting SIP configuration items
- */
- local class SipConfigurationItemVisitor extends AsteriskSCF::System::Configuration::V1::ConfigurationItemVisitor
- {
- };
+/**
+ * General SIP configuration group that contains general items related to the SIP component as a whole
+ */
+class SipGeneralGroup extends SipConfigurationGroup
+{
+};
- /**
- * Generic SIP configuration item
- */
- ["visitor:SipConfigurationItemVisitor"] class SipConfigurationItem extends AsteriskSCF::System::Configuration::V1::ConfigurationItem
- {
- };
+/**
+ * Local visitor class for visiting SIP configuration items
+ */
+local class SipConfigurationItemVisitor extends AsteriskSCF::System::Configuration::V1::ConfigurationItemVisitor
+{
+};
- /**
- * SIP Endpoint group, used to configure an endpoint
- */
- class SipEndpointGroup extends SipConfigurationGroup
- {
- /**
- * Name of the endpoint being configured
- */
- string name;
- };
+/**
+ * Generic SIP configuration item
+ */
+["visitor:SipConfigurationItemVisitor"] class SipConfigurationItem extends AsteriskSCF::System::Configuration::V1::ConfigurationItem
+{
+};
- /**
- * SIP Domain group, used to configure a specific domain
- */
- class SipDomainGroup extends SipConfigurationGroup
- {
- /**
- * Name of the domain being configured
- */
- string domain;
- };
+/**
+ * SIP Endpoint group, used to configure an endpoint
+ */
+class SipEndpointGroup extends SipConfigurationGroup
+{
+ /**
+ * Name of the endpoint being configured
+ */
+ string name;
+};
- /**
- * Domain item for bindings
- */
- class SipBindingsItem extends SipConfigurationItem
- {
- /**
- * Addresses the domain is reachable over
- */
- Ice::StringSeq bindings;
- };
+/**
+ * SIP Domain group, used to configure a specific domain
+ */
+class SipDomainGroup extends SipConfigurationGroup
+{
+ /**
+ * Name of the domain being configured
+ */
+ string domain;
+};
- /**
- * Generic SIP transport group that others derive from, primarily created since each one has
- * the requirement of a name
- */
- class SipTransportGroup extends SipConfigurationGroup
- {
- /**
- * Name of the specific transport. Since each one may have multiple transports this differentiates
- * them.
- */
- string name;
- };
+/**
+ * Domain item for bindings
+ */
+class SipBindingsItem extends SipConfigurationItem
+{
+ /**
+ * Addresses the domain is reachable over
+ */
+ Ice::StringSeq bindings;
+};
- /**
- * Host information configuration item
- */
- class SipHostItem extends SipConfigurationItem
- {
- /**
- * String containing the IP address or string address
- */
- string host;
-
- /**
- * Port number
- */
- int port;
- };
+/**
+ * Generic SIP transport group that others derive from, primarily created since each one has
+ * the requirement of a name
+ */
+class SipTransportGroup extends SipConfigurationGroup
+{
+ /**
+ * Name of the specific transport. Since each one may have multiple transports this differentiates
+ * them.
+ */
+ string name;
+};
- /**
- * Source transport address configuration item
- */
- class SipSourceTransportAddressItem extends SipHostItem
- {
- };
+/**
+ * Host information configuration item
+ */
+class SipHostItem extends SipConfigurationItem
+{
+ /**
+ * String containing the IP address or string address
+ */
+ string host;
- /**
- * Target destination address configuration item
- */
- class SipTargetDestinationAddressItem extends SipHostItem
- {
- };
+ /**
+ * Port number
+ */
+ int port;
+};
- /**
- * Crypto certificate configuration item
- */
- class SipCryptoCertificateItem extends SipConfigurationItem
- {
- /**
- * Full location of certificate authority file
- */
- string certificateAuthority;
-
- /**
- * Full location of certificate file
- */
- string certificate;
-
- /**
- * Full location of private key file
- */
- string privateKey;
-
- /**
- * Password to open private key file
- */
- string privateKeyPassword;
- };
+/**
+ * Source transport address configuration item
+ */
+class SipSourceTransportAddressItem extends SipHostItem
+{
+};
- /**
- * Crypto requirement configuration item
- */
- class SipCryptoRequirementsItem extends SipConfigurationItem
- {
- /**
- * Require verification of server certificate
- */
- bool requireVerifiedServer = false;
-
- /**
- * Require verification of client certificate
- */
- bool requireVerifiedClient = false;
-
- /**
- * Require client certificate be present
- */
- bool requireClientCertificate = false;
- };
+/**
+ * Target destination address configuration item
+ */
+class SipTargetDestinationAddressItem extends SipHostItem
+{
+};
- /**
- * TLS protocol methods
- */
- enum TLSProtocolMethod
- {
- PROTOCOLMETHODUNSPECIFIED,
- PROTOCOLMETHODTLSV1,
- PROTOCOLMETHODSSLV2,
- PROTOCOLMETHODSSLV3,
- PROTOCOLMETHODSSLV23,
- };
+/**
+ * Crypto certificate configuration item
+ */
+class SipCryptoCertificateItem extends SipConfigurationItem
+{
+ /**
+ * Full location of certificate authority file
+ */
+ string certificateAuthority;
- /**
- * General crypto configuration item
- */
- class SipCryptoItem extends SipConfigurationItem
- {
- /**
- * TLS protocol method to use
- */
- TLSProtocolMethod protocolMethod = PROTOCOLMETHODUNSPECIFIED;
-
- /**
- * Supported ciphers (OpenSSL format)
- */
- string supportedCiphers;
-
- /**
- * Server name
- */
- string serverName;
-
- /**
- * TLS negotiation timeout in seconds
- */
- int timeout = 0;
- };
+ /**
+ * Full location of certificate file
+ */
+ string certificate;
- /**
- * Routing service configuration item
- */
- class SipRoutingItem extends SipConfigurationItem
- {
- /**
- * Name of the routing service to use
- */
- string routingServiceName;
- };
+ /**
+ * Full location of private key file
+ */
+ string privateKey;
- /**
- * RTP Media service configuration item
- */
- class SipRTPMediaServiceItem extends SipConfigurationItem
- {
- /**
- * Name of the RTP media service to use
- */
- string mediaServiceName;
-
- /**
- * Whether to choose an IPv6 RTP media service or not
- */
- bool requireIPv6 = false;
- };
+ /**
+ * Password to open private key file
+ */
+ string privateKeyPassword;
+};
- /**
- * Signaling NAT configuration item
- */
- class SipSignalingNATItem extends SipConfigurationItem
- {
- /**
- * Boolean for whether STUN is enabled
- */
- bool stun;
- };
+/**
+ * Crypto requirement configuration item
+ */
+class SipCryptoRequirementsItem extends SipConfigurationItem
+{
+ /**
+ * Require verification of server certificate
+ */
+ bool requireVerifiedServer = false;
- /**
- * Media NAT configuration item
- */
- class SipMediaNATItem extends SipConfigurationItem
- {
- /**
- * Boolean for whether symmetric RTP is enabled
- */
- bool symmetricRTP;
-
- /**
- * Boolean for whether STUN is enabled
- */
- bool stun;
-
- /**
- * Boolean for whether ICE is enabled
- */
- bool interactiveconnectivityestablishment;
-
- /**
- * Boolean for whether TURN is enabled
- */
- bool turn;
- };
+ /**
+ * Require verification of client certificate
+ */
+ bool requireVerifiedClient = false;
- /**
- * Access control lists item
- */
- class SipACLItem extends SipConfigurationItem
- {
- /**
- * Name of this ACL item. A user may want to have multiple.
- */
- string name;
-
- /**
- * Ranges of allowed hosts
- */
- Ice::StringSeq allowRanges;
-
- /**
- * Specific allowed hosts
- */
- Ice::StringSeq allowHosts;
-
- /**
- * Ranges of denied hosts
- */
- Ice::StringSeq denyRanges;
-
- /**
- * Specific denied hosts
- */
- Ice::StringSeq denyHosts;
- };
+ /**
+ * Require client certificate be present
+ */
+ bool requireClientCertificate = false;
+};
- /**
- * Allowable call directions
- */
- enum SipAllowableCallDirection
- {
- Disabled,
- Inbound,
- Outbound,
- Both,
- };
+/**
+ * TLS protocol methods
+ */
+enum TLSProtocolMethod
+{
+ PROTOCOLMETHODUNSPECIFIED,
+ PROTOCOLMETHODTLSV1,
+ PROTOCOLMETHODSSLV2,
+ PROTOCOLMETHODSSLV3,
+ PROTOCOLMETHODSSLV23,
+};
+
+/**
+ * General crypto configuration item
+ */
+class SipCryptoItem extends SipConfigurationItem
+{
+ /**
+ * TLS protocol method to use
+ */
+ TLSProtocolMethod protocolMethod = PROTOCOLMETHODUNSPECIFIED;
+
+ /**
+ * Supported ciphers (OpenSSL format)
+ */
+ string supportedCiphers;
+
+ /**
+ * Server name
+ */
+ string serverName;
+
+ /**
+ * TLS negotiation timeout in seconds
+ */
+ int timeout = 0;
+};
+
+/**
+ * SRTP Crypto Key information.
+ */
+class SRTPCryptoKey
+{
+ /**
+ * The cipher suite name.
+ */
+ string suite;
+
+ /**
+ * The key information for the suite.
+ */
+ string cryptoKey;
+};
+
+sequence<SRTPCryptoKey> SRTPCryptoKeySeq;
+
+/**
+ * SRTP crypto configuration.
+ */
+class SRTPCryptoItem extends SipConfigurationItem
+{
+
+ /**
+ * Most of the time the following values will be false. They can be overriden to be true however.
+ */
+ bool enableAuthentication;
+ bool enableEncryption;
+
+ /**
+ * The suites and keys supported by this endpoint.
+ */
+ SRTPCryptoKeySeq cryptoKeys;
+};
+
+/**
+ * Routing service configuration item
+ */
+class SipRoutingItem extends SipConfigurationItem
+{
+ /**
+ * Name of the routing service to use
+ */
+ string routingServiceName;
+};
+
+/**
+ * RTP Media service configuration item
+ */
+class SipRTPMediaServiceItem extends SipConfigurationItem
+{
+ /**
+ * Name of the RTP media service to use
+ */
+ string mediaServiceName;
+
+ /**
+ * Whether to choose an IPv6 RTP media service or not
+ */
+ bool requireIPv6 = false;
+};
+
+/**
+ * Signaling NAT configuration item
+ */
+class SipSignalingNATItem extends SipConfigurationItem
+{
+ /**
+ * Boolean for whether STUN is enabled
+ */
+ bool stun;
+};
+
+/**
+ * STUN server to be used by Sip signalling.
+ */
+class SipSignalingSTUNHostItem extends SipConfigurationItem
+{
+ /**
+ * Address for the STUN server.
+ **/
+ string address;
+
+ /**
+ * Port for the STUN host.
+ */
+ int port;
+};
+
+/**
+ * Media NAT configuration item
+ */
+class SipMediaNATItem extends SipConfigurationItem
+{
+ /**
+ * Boolean for whether ICE is enabled
+ */
+ bool enableICE;
+
+ /**
+ * Boolean for whether TURN is enabled
+ */
+ bool enableTURN;
+};
+
+/**
+ * Access control lists item
+ */
+class SipACLItem extends SipConfigurationItem
+{
+ /**
+ * Name of this ACL item. A user may want to have multiple.
+ */
+ string name;
+
+ /**
+ * Ranges of allowed hosts
+ */
+ Ice::StringSeq allowRanges;
+ /**
+ * Specific allowed hosts
+ */
+ Ice::StringSeq allowHosts;
+
+ /**
+ * Ranges of denied hosts
+ */
+ Ice::StringSeq denyRanges;
+
+ /**
+ * Specific denied hosts
+ */
+ Ice::StringSeq denyHosts;
+};
+
+/**
+ * Allowable call directions
+ */
+enum SipAllowableCallDirection
+{
+ Disabled,
+ Inbound,
+ Outbound,
+ Both,
+};
+
+/**
+ * Allowable call direction item
+ */
+class SipAllowableCallDirectionItem extends SipConfigurationItem
+{
+ /**
+ * What direction is permitted for this endpoint
+ */
+ SipAllowableCallDirection callDirection = Both;
+};
+
+/**
+* Allowable media format item
+*/
+class SipMediaFormatItem extends SipConfigurationItem
+{
/**
- * Allowable call direction item
+ * Name of the media format
*/
- class SipAllowableCallDirectionItem extends SipConfigurationItem
- {
- /**
- * What direction is permitted for this endpoint
- */
- SipAllowableCallDirection callDirection = Both;
- };
+ string name;
/**
- * Allowable media format item
+ * Sample rate for frames
+ *
+ * This is specified in Hz and has a default value of 8000.
*/
- class SipMediaFormatItem extends SipConfigurationItem
- {
- /**
- * Name of the media format
- */
- string name;
-
- /**
- * Sample rate for frames
- *
- * This is specified in Hz and has a default value of 8000.
- */
- int sampleRate = 8000;
-
- /**
- * Amount of audio in frames
- *
- * This is specified in
- */
- int frameSize;
-
- /**
- * Any format specific parameters
- */
- Ice::StringSeq formatSpecific;
- };
+ int sampleRate = 8000;
/**
- * User agent presentation configuration item
+ * Amount of audio in frames
+ *
+ * This is specified in
*/
- class SipUserAgentItem extends SipConfigurationItem
- {
- /**
- * String that our user agent should appear as
- */
- string userAgent;
- };
+ int frameSize;
/**
- * Transport information configuration item
+ * Format specific attributes.
*/
- class SipEndpointTransportItem extends SipConfigurationItem
- {
- /**
- * What directions to require secure transport in
- */
- SipAllowableCallDirection secureTransport = Disabled;
- };
+ Ice::StringSeq formatSpecific;
+};
+/**
+ * User agent presentation configuration item
+ */
+class SipUserAgentItem extends SipConfigurationItem
+{
/**
- * Methods of transporting DTMF via SIP
+ * String that our user agent should appear as
*/
- enum SipDTMFOption
- {
- Inband,
- RFC4733,
- INFO,
- };
+ string userAgent;
+};
+/**
+ * Transport information configuration item
+ */
+class SipEndpointTransportItem extends SipConfigurationItem
+{
/**
- * Denotes the method by which we will send DTMF to an endpoint
+ * What directions to require secure transport in
*/
- class SipDTMFItem extends SipConfigurationItem
- {
- SipDTMFOption dtmf = RFC4733;
- };
+ SipAllowableCallDirection secureTransport = Disabled;
+};
- /**
- * Group of configuration items related to a UDP transport
- */
- class SipUDPTransportGroup extends SipTransportGroup
- {
- };
+/**
+ * Group of configuration items related to a UDP transport
+ */
+class SipUDPTransportGroup extends SipTransportGroup
+{
+};
- /**
- * Group of configuration items related to a TCP transport
- */
- class SipTCPTransportGroup extends SipTransportGroup
- {
- };
+/**
+ * Group of configuration items related to a TCP transport
+ */
+class SipTCPTransportGroup extends SipTransportGroup
+{
+};
- /**
- * Group of configuration items related to a TLS transport
- */
- class SipTLSTransportGroup extends SipTransportGroup
- {
- };
+/**
+ * Group of configuration items related to a TLS transport
+ */
+class SipTLSTransportGroup extends SipTransportGroup
+{
+};
+
+/**
+ * Group of configuration items related to a STUN enabled transport
+ */
+class SipSTUNTransportGroup extends SipTransportGroup
+{
+};
+
++/**
++ * Methods of transporting DTMF via SIP
++ */
++enum SipDTMFOption
++{
++ Inband,
++ RFC4733,
++ INFO,
++};
++
++/**
++ * Denotes the method by which we will send DTMF to an endpoint
++ */
++class SipDTMFItem extends SipConfigurationItem
++{
++ SipDTMFOption dtmf = RFC4733;
++};
+
}; /* module V1 */
}; /* module SipSessionManager */
diff --cc src/PJSipSessionModule.cpp
index 0ceb682,062f997..44a1469
--- a/src/PJSipSessionModule.cpp
+++ b/src/PJSipSessionModule.cpp
@@@ -20,11 -20,11 +20,13 @@@
#include "SipSession.h"
#include "PJSipManager.h"
#include "SipStateReplicator.h"
+ #include "SipTelephonyEventSource.h"
#include <IceUtil/UUID.h>
+ #include <boost/lexical_cast.hpp>
+#include <boost/lexical_cast.hpp>
+
#include <AsteriskSCF/Core/Endpoint/EndpointIf.h>
#include <AsteriskSCF/Core/Routing/RoutingIf.h>
#include <AsteriskSCF/SessionCommunications/SessionCommunicationsIf.h>
@@@ -99,18 -95,16 +101,18 @@@ void PJSipSessionModInfo::updateSession
boost::unique_lock<boost::shared_mutex> lock(mLock);
if (mSession)
{
- mSessionState->mEndpointName = mSession->getEndpoint()->getName();
- mSessionState->mSessionObjectId = mSession->getSessionProxy()->ice_getIdentity();
- mSessionState->mMediaSessionObjectId = mSession->getMediaSessionProxy()->ice_getIdentity();
- mSessionState->mSources = mSession->getMediaSources();
- mSessionState->mSinks = mSession->getMediaSinks();
- mSessionState->mRTPMediaSessions = mSession->getRTPMediaSessions();
- mSessionState->mListeners = mSession->getListeners();
+ mSessionState->endpointName = mSession->getEndpoint()->getName();
+ mSessionState->sessionObjectId = mSession->getSessionProxy()->ice_getIdentity();
+ mSessionState->mediaSessionObjectId = mSession->getMediaSessionProxy()->ice_getIdentity();
+ mSessionState->sessionControllerObjectId = mSession->getOurSessionControllerProxy()->ice_getIdentity();
- mSessionState->sources = mSession->getSources();
- mSessionState->sinks = mSession->getSinks();
++ mSessionState->sources = mSession->getMediaSources();
++ mSessionState->sinks = mSession->getMediaSinks();
+ mSessionState->rtpMediaSessions = mSession->getRTPMediaSessions();
+ mSessionState->listeners = mSession->getListeners();
+ mSessionState->streams = mSession->getStreams();
try
{
- mSessionState->mBridge = mSession->getBridge();
+ mSessionState->bridge = mSession->getBridge();
}
catch (...)
{
@@@ -459,18 -366,19 +461,20 @@@ protected
return Complete;
}
- mSession->setInviteSession(mInv);
- mSession->setDialog(mInv->dlg);
+ mSession->setInviteSession(mInv);
+ mSession->setDialog(mInv->dlg);
+
+ mSession->setTelephonyEventSourcesAndSinks(mCaller->getConfig());
// Create an SDP offer or answer
- const pjmedia_sdp_session *remote_sdp = NULL;
+ const pjmedia_sdp_session *remote_sdp = NULL;
pjmedia_sdp_session *sdp;
+ StreamInformationDict streams;
- if (!mInv->neg || (pjmedia_sdp_neg_get_neg_remote(mInv->neg, &remote_sdp) != PJ_SUCCESS))
+ if (!mInv->neg || (pjmedia_sdp_neg_get_neg_remote(mInv->neg, &remote_sdp) != PJ_SUCCESS))
{
// No SDP was present in the INVITE so we need to create an offer
- sdp = mSession->createSDPOffer();
+ sdp = mSession->createSDPOffer(StreamInformationDict(), streams);
}
else
{
@@@ -1057,6 -957,218 +1061,219 @@@ void PJSipSessionModule::handleRefer(pj
enqueueSessionWork(new HandleReferOperation(inv, tsx, tdata, replaces_param, to_tag_param, from_tag_param, target, session, mSessionRouter, mModule.id), inv);
}
+ class HandleInfoDTMFOperation : public SipQueueableOperation
+ {
+ public:
+ HandleInfoDTMFOperation(
+ pjsip_inv_session *inv,
+ pjsip_transaction *tsx,
+ pjsip_tx_data *tdata,
+ const SipSessionPtr& session,
+ char dtmf,
+ int duration)
+ : mInv(inv), mTsx(tsx), mTdata(tdata), mSession(session), mDTMF(dtmf), mDuration(duration) { }
+
- SuspendableWorkResult initial(const SuspendableWorkListenerPtr& workListener)
++ SuspendableWorkResult initial(const SuspendableWorkListenerPtr&)
+ {
+ TelephonyEventPtr event;
+ if (mDTMF == '!')
+ {
+ event = new FlashEvent();
+ }
+ else
+ {
+ event = new EndDTMFEvent(mDTMF, mDuration);
+ }
+ SipTelephonyEventSourcePtr source = mSession->getSipTelephonyEventSource();
+ source->distributeToSinks(event);
+ pjsip_dlg_send_response(mInv->dlg, mTsx, mTdata);
++ return Complete;
+ }
+
+ private:
+ pjsip_inv_session *mInv;
+ pjsip_transaction *mTsx;
+ pjsip_tx_data *mTdata;
+ SipSessionPtr mSession;
+ char mDTMF;
+ int mDuration;
+ };
+
+ bool PJSipSessionModule::isDTMF(pjsip_rx_data *rdata)
+ {
+ // We should determine how to handle matters based on the "Info-Package" header,
+ // but DTMF via INFO existed long before the Info-Package RFC was published.
+ // An official document defining DTMF over INFO is still in the draft stage as
+ // of July 2011. We will thus use the Info-Package or the Content-Type header
+ // to determine if this is DTMF.
+
+ pj_str_t str_info_pkg;
+ pj_cstr(&str_info_pkg, "Info-Package");
+ pjsip_generic_string_hdr *info_package =
+ static_cast<pjsip_generic_string_hdr *>(pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_info_pkg, NULL));
+
+ pj_str_t dtmf;
+ pj_cstr(&dtmf, "dtmf");
+ if (info_package && !pj_strcmp(&info_package->hvalue, &dtmf))
+ {
+ lg(Debug) << "Info-Package header suggests DTMF";
+ return true;
+ }
+
+ lg(Debug) << "No Info-Package header. Checking Content-Type";
+ //No biggie. I just said that not many are actually going to use Info-Package.
+ //Check if the content-type is "application/dtmf-relay"
+ pj_str_t application;
+ pj_str_t dtmf_relay;
+ pj_cstr(&application, "application");
+ pj_cstr(&dtmf_relay, "dtmf-relay");
+ if (!pj_strcmp(&rdata->msg_info.ctype->media.type, &application) &&
+ !pj_strcmp(&rdata->msg_info.ctype->media.subtype, &dtmf_relay))
+ {
+ lg(Debug) << "Content-Type is application/dtmf-relay";
+ return true;
+ }
+
+ return false;
+ }
+
+ class DTMFInfoException : public std::exception
+ {
+ public:
+ DTMFInfoException(const std::string& msg)
+ : mMsg(msg) { }
+
+ ~DTMFInfoException() throw() { }
+
+ const char *what() const throw()
+ {
+ return mMsg.c_str();
+ }
+ private:
+ const std::string mMsg;
+ };
+
+ char PJSipSessionModule::getDTMFInfoSignal(pjsip_msg_body *body)
+ {
+ char *data = static_cast<char *>(body->data);
+ std::string dataStr(data, body->len);
+ std::string signalStr("Signal=");
+ std::string altSignalStr("d=");
+ std::stringstream err;
+
+ size_t signalPos = dataStr.find(signalStr);
+ if (signalPos == std::string::npos)
+ {
+ signalPos = dataStr.find(altSignalStr);
+ if (signalPos == std::string::npos)
+ {
+ err << "Couldn't find " << signalStr << " or " << altSignalStr << " in DTMF INFO body";
+ throw DTMFInfoException(err.str());
+ }
+ }
+
+ //The pending draft for DTMF via INFO mandates a single space after
+ //'Signal=' but pre-existing implementations do not necessarily have
+ //any spaces and may do something silly like have multiple spaces.
+ size_t signalValuePos = dataStr.find_first_not_of(' ', signalPos + signalStr.size());
+ char retSignal = dataStr.at(signalValuePos);
+
+ //'!' is used by some endpoints to indicate a flash
+ std::string validDTMF("1234567890#*ABCD!");
+ if (!validDTMF.find(retSignal))
+ {
+ err << "Signal '" << retSignal << "' is not a valid DTMF signal";
+ throw DTMFInfoException(err.str());
+ }
+
+ return dataStr.at(signalValuePos);
+ }
+
+ int PJSipSessionModule::getDTMFInfoDuration(pjsip_msg_body *body)
+ {
+ char *data = static_cast<char *>(body->data);
+ std::string dataStr(data, body->len);
+ std::string durationStr("Duration=");
+ std::stringstream err;
+
+ size_t durationPos = dataStr.find(durationStr);
+ if (durationPos == std::string::npos)
+ {
+ err << "Couldn't find " << durationStr << " in DTMF INFO body";
+ throw DTMFInfoException(err.str());
+ }
+
+ //The pending draft for DTMF via INFO mandates a single space after
+ //'Duration=' but pre-existing implementations do not necessarily have
+ //any spaces and may do something silly like have multiple spaces.
+ size_t durationValuePos = durationPos + durationStr.size();
+ size_t durationBegin = dataStr.find_first_not_of(' ', durationValuePos);
+ size_t durationEnd = dataStr.find_first_not_of(" 1234567890", durationValuePos);
+ size_t durationSize = durationEnd - durationBegin;
+ std::string duration(dataStr.substr(durationBegin, durationSize));
+
+ int retDuration;
+ try
+ {
+ retDuration = boost::lexical_cast<int>(duration);
+ }
+ catch (const boost::bad_lexical_cast& ex)
+ {
+ throw DTMFInfoException(ex.what());
+ }
+
+ return retDuration;
+ }
+
+ void PJSipSessionModule::handleInfo(pjsip_inv_session *inv, pjsip_rx_data *rdata)
+ {
+
+ //Before we do anything, we need to figure out if we should even be handling this
+ //at all. If we don't have a telephony event source to handle this, then just
+ //ignore.
+ PJSipSessionModInfo *sessionModInfo = static_cast<PJSipSessionModInfo*>(inv->mod_data[mModule.id]);
+ SipSessionPtr session = sessionModInfo->getSessionPtr();
+
+ if (!session->isTelephonyEventSource())
+ {
+ //RFC 2976 Section 2.2 states that if we receive an INFO request with a body
+ //type we understand but have no intention of processing, we should respond
+ //with a 200 OK
+
+ pjsip_dlg_respond(inv->dlg, rdata, 200, NULL, NULL, NULL);
+ return;
+ }
+
+ // The only INFO package we handle for now is "dtmf". We need to verify that
+ // what we have received is actually DTMF and not some other INFO package.
+
+ if (!isDTMF(rdata))
+ {
+ return;
+ }
+
+ char dtmf;
+ int duration;
+ //It's DTMF! Grab the info we need! HA THAT WAS A PUN YOU JERK!
+ try
+ {
+ dtmf = getDTMFInfoSignal(rdata->msg_info.msg->body);
+ duration = getDTMFInfoDuration(rdata->msg_info.msg->body);
+ }
+ catch (const DTMFInfoException& ex)
+ {
+ lg(Error) << "Error processing DTMF INFO: " << ex.what();
+ pjsip_dlg_respond(inv->dlg, rdata, 400, NULL, NULL, NULL);
+ return;
+ }
+
+ pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+ pjsip_tx_data *tdata;
+ pjsip_dlg_create_response(inv->dlg, rdata, 200, NULL, &tdata);
+ enqueueSessionWork(new HandleInfoDTMFOperation(inv, tsx, tdata, session, dtmf, duration), inv);
+ return;
+ }
+
pj_bool_t PJSipSessionModule::on_rx_request(pjsip_rx_data *rdata)
{
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
diff --cc src/PJSipSessionModule.h
index 3f9ed89,6337e57..e7f6e62
--- a/src/PJSipSessionModule.h
+++ b/src/PJSipSessionModule.h
@@@ -143,9 -121,10 +143,13 @@@ private
void handleNewInvite(pjsip_rx_data *rdata);
void handleInviteResponse(pjsip_inv_session *inv, pjsip_rx_data *rdata, pjsip_dialog *dlg);
void handleRefer(pjsip_inv_session *inv, pjsip_rx_data *rdata);
+ void handleInfo(pjsip_inv_session *inv, pjsip_rx_data *rdata);
+ bool isDTMF(pjsip_rx_data *rdata);
+ char getDTMFInfoSignal(pjsip_msg_body *body);
+ int getDTMFInfoDuration(pjsip_msg_body *body);
+ bool checkAuth(pjsip_rx_data *rdata, pjsip_inv_session *inv, AsteriskSCF::SIP::ExtensionPoint::V1::RequestType type);
+ void getURIParams(pjsip_uri *uri, AsteriskSCF::SIP::ExtensionPoint::V1::ParamDict ¶ms);
+ void createAuthManager(pjsip_endpoint *endpt);
pjsip_inv_callback mInvCallback;
pjsip_ua_init_param mUaParam;
const std::string mName;
diff --cc src/SipConfiguration.cpp
index 65a40a2,eb8bf6f..d6a342a
--- a/src/SipConfiguration.cpp
+++ b/src/SipConfiguration.cpp
@@@ -215,15 -209,10 +215,19 @@@ class EndpointConfigHelper : public boo
mUpdates.push_back(boost::bind(&EndpointConfigHelper::updateTransport, mConfig, transport));
};
+ void visitSipMediaNATItem(const SipMediaNATItemPtr& item)
+ {
+ mUpdates.push_back(boost::bind(&EndpointConfigHelper::updateMediaNATOptions, mConfig, item));
+ }
+
+ void visitSipSignalingNATItem(const SipSignalingNATItemPtr& item)
+ {
+ mUpdates.push_back(boost::bind(&EndpointConfigHelper::updateSignalingNAT, mConfig, item));
+ }
+ void visitSipDTMFItem(const SipDTMFItemPtr& dtmf)
+ {
+ mUpdates.push_back(boost::bind(&EndpointConfigHelper::updateDTMF, mConfig, dtmf));
+ };
private:
@@@ -304,31 -293,10 +308,35 @@@ public
mEndpoint->setSecureTransport(translateCallDirection(transport->secureTransport));
}
+ void updateSRTP(const SRTPCryptoItemPtr& srtpOptions)
+ {
+ SipEndpointMediaSRTPConfig srtpConfig(mEndpoint->getConfig().srtpConfig);
+ srtpConfig.enableEncryption = srtpOptions->enableEncryption;
+ srtpConfig.enableAuthentication = srtpOptions->enableAuthentication;
+ srtpConfig.cryptoKeys.clear();
+ if (!srtpOptions->cryptoKeys.empty())
+ {
+ CryptoKey newKey;
+ newKey.suite = srtpOptions->cryptoKeys[0]->suite;
+ newKey.cryptoKey = srtpOptions->cryptoKeys[0]->cryptoKey;
+ srtpConfig.cryptoKeys.push_back(newKey);
+ }
+ mEndpoint->setSRTPOptions(srtpConfig);
+ }
+
+ void updateMediaNATOptions(const SipMediaNATItemPtr& item)
+ {
+ mEndpoint->setMediaNATOptions(item->enableICE, item->enableTURN);
+ }
+
+ void updateSignalingNAT(const SipSignalingNATItemPtr& item)
+ {
+ mEndpoint->setSignalingNATOptions(item->stun);
+ }
+ void updateDTMF(const SipDTMFItemPtr& dtmf)
+ {
+ mEndpoint->setDTMFMethod(dtmf->dtmf);
+ }
void updated(const UpdateCommandList& updates)
{
diff --cc src/SipEndpoint.cpp
index 624da0c,a37281f..f8e0cf5
--- a/src/SipEndpoint.cpp
+++ b/src/SipEndpoint.cpp
@@@ -368,10 -342,7 +373,10 @@@ AsteriskSCF::SessionCommunications::V1:
}
SipSessionPtr session = new SipSession(mImplPriv->mAdapter, this, destination, listener, mImplPriv->mManager,
- mImplPriv->mServiceLocator, mImplPriv->mReplica, mImplPriv->mConfig.sessionConfig.rtpOverIPv6, true,
- mImplPriv->mServiceLocator, mImplPriv->mReplica, true, mImplPriv->mConfig);
++ mImplPriv->mServiceLocator, mImplPriv->mReplica, mImplPriv->mConfig.sessionConfig.rtpOverIPv6, true, mImplPriv->mConfig,
+ NATEndpointOptions(mImplPriv->mConfig.sessionConfig.rtpOverICE, mImplPriv->mConfig.sessionConfig.rtpICEIncludeTURN,
+ mImplPriv->mConfig.transportConfig.enableNAT));
+
mImplPriv->mSessions.push_back(session);
std::cout << "And now we're returing a session proxy..." << std::endl;
return session->getSessionProxy();
@@@ -380,11 -351,7 +385,11 @@@
AsteriskSCF::SipSessionManager::SipSessionPtr SipEndpoint::createSession(const std::string& destination)
{
SipSessionPtr session = new SipSession(mImplPriv->mAdapter, this, destination, 0, mImplPriv->mManager,
- mImplPriv->mServiceLocator, mImplPriv->mReplica, mImplPriv->mConfig.sessionConfig.rtpOverIPv6, false,
- mImplPriv->mServiceLocator, mImplPriv->mReplica, false, mImplPriv->mConfig);
++ mImplPriv->mServiceLocator, mImplPriv->mReplica, mImplPriv->mConfig.sessionConfig.rtpOverIPv6, false, mImplPriv->mConfig,
+ NATEndpointOptions(mImplPriv->mConfig.sessionConfig.rtpOverICE, mImplPriv->mConfig.sessionConfig.rtpICEIncludeTURN,
+ mImplPriv->mConfig.transportConfig.enableNAT)
+ );
+
mImplPriv->mSessions.push_back(session);
return session;
}
@@@ -396,10 -362,8 +401,10 @@@ AsteriskSCF::SipSessionManager::SipSess
const AsteriskSCF::Media::V1::StreamSourceSeq& sources,
const AsteriskSCF::Media::V1::StreamSinkSeq& sinks)
{
- SipSessionPtr session = new SipSession(mImplPriv->mAdapter, this, destination, sessionid, mediaid, mediasessions,
- sources, sinks, mImplPriv->mManager, mImplPriv->mServiceLocator, mImplPriv->mReplica, false, mImplPriv->mConfig);
+ SipSessionPtr session = new SipSession(mImplPriv->mAdapter, this, destination, sessionid, controllerid, mediaid, mediasessions,
- sources, sinks, mImplPriv->mManager, mImplPriv->mServiceLocator, mImplPriv->mReplica, false,
++ sources, sinks, mImplPriv->mManager, mImplPriv->mServiceLocator, mImplPriv->mReplica, false, mImplPriv->mConfig,
+ NATEndpointOptions(mImplPriv->mConfig.sessionConfig.rtpOverICE, mImplPriv->mConfig.sessionConfig.rtpICEIncludeTURN,
+ mImplPriv->mConfig.transportConfig.enableNAT));
mImplPriv->mSessions.push_back(session);
return session;
}
diff --cc src/SipEndpoint.h
index a0657d5,6991acb..d4afefc
--- a/src/SipEndpoint.h
+++ b/src/SipEndpoint.h
@@@ -157,37 -142,9 +158,40 @@@ public
// Whether we are using IPv6 for media transport
// or not.
bool rtpOverIPv6;
+ // The method by which we will transmit
+ // DTMF to an endpoint
+ AsteriskSCF::Configuration::SipSessionManager::V1::SipDTMFOption dtmf;
+ //
+ // Enables ICE transport.
+ //
+ bool rtpOverICE;
+ //
+ // Enables TURN as a viable mechanism for traversing NATs.
+ //
+ bool rtpICEIncludeTURN;
+
+ SipEndpointSessionConfig() :
+ rtpOverIPv6(false),
+ rtpOverICE(false),
+ rtpICEIncludeTURN(false)
+ {
+ }
+};
+
+class CryptoKey
+{
+public:
+ std::string suite;
+ std::string cryptoKey;
+};
+typedef std::vector<CryptoKey> CryptoKeys;
+
+class SipEndpointMediaSRTPConfig
+{
+public:
+ bool enableAuthentication;
+ bool enableEncryption;
+ CryptoKeys cryptoKeys;
};
/**
@@@ -330,10 -280,8 +334,11 @@@ public
/**
* API call which returns the stream topology to be used for an SDP offer.
*/
- StreamTopologyMap getStreamTopology();
+ void setMediaNATOptions(bool useICE, bool useTURN);
+ void setSRTPOptions(const SipEndpointMediaSRTPConfig& srtpConfig);
+ void setSignalingNATOptions(bool enable);
+ AsteriskSCF::Media::V1::StreamInformationDict getStreamTopology();
+ void setDTMFMethod(AsteriskSCF::Configuration::SipSessionManager::V1::SipDTMFOption dtmf);
private:
/**
diff --cc src/SipSession.cpp
index 1e494f6,a50c158..aa2c3f5
mode 100755,100644..100755
--- a/src/SipSession.cpp
+++ b/src/SipSession.cpp
@@@ -295,30 -269,21 +297,42 @@@ public
pjmedia_sdp_session *mSDP;
/**
- * Whether the SDP has been finalized or not.
+ * Streams present on the session.
+ */
+ StreamInformationDict mStreams;
+
+ /**
+ * Newly added streams that are awaiting a response from the remote party on acceptance.
*/
- bool mSDPFinalized;
+ StreamInformationDict mNewStreams;
+
+ /**
+ * A proxy to the SessionController we are controlling.
+ */
+ AsteriskSCF::SessionCommunications::V1::SessionControllerPrx mSessionController;
+
+ /**
+ * A pointer to the callback class used to send a response to the SessionController with which streams
+ * have been added.
+ */
+ AsteriskSCF::SessionCommunications::V1::AMD_SessionController_addStreamsPtr mAddStreamsCb;
+
+ /**
+ * A mapping for requested stream identifier to our stream identifier.
+ */
+ std::map<std::string, std::string> mAddStreamsMapping;
+
+ SipTelephonyEventSinkPtr mEventSink;
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkPrx mEventSinkPrx;
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq mExternalEventSinks;
+
+ SipTelephonyEventSourcePtr mEventSource;
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx mEventSourcePrx;
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourceSeq mExternalEventSources;
};
/**
@@@ -711,50 -357,29 +725,64 @@@ void SipSession::initializePJSIPStructs
setInviteSession(inviteSession);
}
+void SipSession::activateIceObjects(const AsteriskSCF::SessionCommunications::ExtensionPoints::V1::SessionCreationHookSeq& hooks)
+{
+ mImplPriv->mSessionProxy =
+ AsteriskSCF::SessionCommunications::V1::SessionPrx::uncheckedCast(mImplPriv->mAdapter->addWithUUID(this));
+
+ AsteriskSCF::SessionCommunications::ExtensionPoints::V1::SessionCreationHookDataPtr initial(new AsteriskSCF::SessionCommunications::ExtensionPoints::V1::SessionCreationHookData(mImplPriv->mSessionProxy, mImplPriv->mListeners));
+
+ AsteriskSCF::SessionCommunications::ExtensionPoints::V1::SessionCreationHookDataPtr in = initial;
+
+ for (AsteriskSCF::SessionCommunications::ExtensionPoints::V1::SessionCreationHookSeq::const_iterator iter = hooks.begin();
+ iter != hooks.end(); ++iter)
+ {
+ AsteriskSCF::System::Hook::V1::HookResult result;
+ AsteriskSCF::SessionCommunications::ExtensionPoints::V1::SessionCreationHookDataPtr out;
+ try
+ {
+ result = (*iter)->modifySession(in, out);
+ }
+ catch (const Ice::Exception& ex)
+ {
+ lg(Warning) << "Exception caught while trying to set a decorator: " << ex.what();
+ //XXX Probably should remove this hook from the list...
+ continue;
+ }
+
+ if (result.status == AsteriskSCF::System::Hook::V1::Succeeded)
+ {
+ lg(Debug) << "Setting our session proxy to be " << out->session->ice_getIdentity().name;
+ mImplPriv->mPublicSessionProxy = out->session;
+ mImplPriv->mListeners.clear();
+ mImplPriv->mListeners = out->listeners;
+ in = out;
+ }
+ }
+}
+ void SipSession::setTelephonyEventSourcesAndSinks(const SipEndpointConfig& config)
+ {
+ if (config.sessionConfig.dtmf == AsteriskSCF::Configuration::SipSessionManager::V1::INFO)
+ {
+ mImplPriv->mEventSink = new SipTelephonyEventSink(mImplPriv->mSessionWork, mImplPriv->mInviteSession);
+ mImplPriv->mEventSinkPrx =
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkPrx::uncheckedCast(mImplPriv->mAdapter->addWithUUID(mImplPriv->mEventSink));
+
+ mImplPriv->mEventSource = new SipTelephonyEventSource(mImplPriv->mSessionWork);
+ mImplPriv->mEventSourcePrx =
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx::uncheckedCast(mImplPriv->mAdapter->addWithUUID(mImplPriv->mEventSource));
+ }
+ }
/**
* Default constructor.
*/
SipSession::SipSession(const Ice::ObjectAdapterPtr& adapter, const SipEndpointPtr& endpoint,
- const std::string& destination, const AsteriskSCF::SessionCommunications::V1::SessionListenerPrx& listener,
- PJSipManager *manager, const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
- const AsteriskSCF::System::Component::V1::ReplicaPtr& replica, bool isUAC,
- const SipEndpointConfig &config)
- : mImplPriv(new SipSessionPriv(adapter, endpoint, destination, manager, serviceLocator, replica))
+ const std::string& destination, const AsteriskSCF::SessionCommunications::V1::SessionListenerPrx& listener,
+ const PJSipManagerPtr& manager, const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
- const AsteriskSCF::System::Component::V1::ReplicaPtr& replica, bool /* ipv6 */, bool isUAC, const NATEndpointOptions& natOptions)
++ const AsteriskSCF::System::Component::V1::ReplicaPtr& replica, bool /* ipv6 */, bool isUAC, const SipEndpointConfig &config,
++ const NATEndpointOptions& natOptions)
+ : mImplPriv(new SipSessionPriv(adapter, endpoint, destination, manager, serviceLocator, replica, natOptions))
{
if (listener != 0)
{
@@@ -782,15 -405,16 +811,16 @@@
* Replica constructor.
*/
SipSession::SipSession(const Ice::ObjectAdapterPtr& adapter, const SipEndpointPtr& endpoint,
- const std::string& destination, const Ice::Identity& sessionid,
+ const std::string& destination, const Ice::Identity& /* sessionid */,
+ const Ice::Identity& controllerid,
const Ice::Identity& mediaid, const AsteriskSCF::Replication::SipSessionManager::V1::RTPMediaSessionSeq& mediasessions,
const AsteriskSCF::Media::V1::StreamSourceSeq& sources, const AsteriskSCF::Media::V1::StreamSinkSeq& sinks,
- PJSipManager *manager, const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
- const AsteriskSCF::System::Component::V1::ReplicaPtr& replica, bool isUAC,
- const SipEndpointConfig &config)
- : mImplPriv(new SipSessionPriv(adapter, endpoint, destination, manager, serviceLocator, replica))
+ const PJSipManagerPtr& manager, const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
- const AsteriskSCF::System::Component::V1::ReplicaPtr& replica, bool isUAC, const NATEndpointOptions& natOptions)
++ const AsteriskSCF::System::Component::V1::ReplicaPtr& replica, bool isUAC, const SipEndpointConfig &config,
++ const NATEndpointOptions& natOptions)
+ : mImplPriv(new SipSessionPriv(adapter, endpoint, destination, manager, serviceLocator, replica, natOptions))
{
- mImplPriv->mSessionProxy =
- AsteriskSCF::SessionCommunications::V1::SessionPrx::uncheckedCast(adapter->add(this, sessionid));
+ activateIceObjects(mImplPriv->mManager->getSessionModule()->getSessionCreationHooks());
mImplPriv->mMediaSession = new SipMediaSession(this);
mImplPriv->mMediaSessionProxy =
@@@ -1539,6 -1134,74 +1569,76 @@@ AsteriskSCF::SessionCommunications::V1:
return mImplPriv->mSessionCookies;
}
+ class GetSources : public SuspendableWork
+ {
+ public:
+ GetSources(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSourcesPtr& cb,
+ const SipSessionPtr& session)
+ : mCB(cb), mSession(session) { }
+
+ SuspendableWorkResult execute(const SuspendableWorkListenerPtr&)
+ {
+ mCB->ice_response(mSession->getSources());
++ return Complete;
+ }
+ private:
+ AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSourcesPtr mCB;
+ SipSessionPtr mSession;
+ };
+
+ void SipSession::getSources_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSourcesPtr& cb,
+ const Ice::Current&)
+ {
+ enqueueSessionWork(new GetSources(cb, this));
+ }
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourceSeq SipSession::getSources()
+ {
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourceSeq sources(mImplPriv->mExternalEventSources);
+ if (mImplPriv->mEventSource)
+ {
+ sources.push_back(mImplPriv->mEventSourcePrx);
+ }
+ return sources;
+ }
+
+ class GetSinks : public SuspendableWork
+ {
+ public:
+ GetSinks(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSinksPtr& cb,
+ const SipSessionPtr& session)
+ : mCB(cb), mSession(session) { }
+
+ SuspendableWorkResult execute(const SuspendableWorkListenerPtr&)
+ {
+ mCB->ice_response(mSession->getSinks());
++ return Complete;
+ }
+ private:
+ AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSinksPtr mCB;
+ SipSessionPtr mSession;
+ };
+
+ void SipSession::getSinks_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSinksPtr& cb,
+ const Ice::Current&)
+ {
+ enqueueSessionWork(new GetSinks(cb, this));
+ }
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq SipSession::getSinks()
+ {
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq sinks(mImplPriv->mExternalEventSinks);
+ if (mImplPriv->mEventSink)
+ {
+ sinks.push_back(mImplPriv->mEventSinkPrx);
+ }
+ return sinks;
+ }
+
/**
* Internal function called to destroy an endpoint. This is controlled by signaling.
*/
diff --cc src/SipSession.h
index fa89ad0,6e8d7a5..10faf86
--- a/src/SipSession.h
+++ b/src/SipSession.h
@@@ -106,17 -100,17 +111,17 @@@ class SipSession : public AsteriskSCF::
{
public:
SipSession(const Ice::ObjectAdapterPtr&, const SipEndpointPtr&, const std::string&,
- const AsteriskSCF::SessionCommunications::V1::SessionListenerPrx&, PJSipManager *manager,
+ const AsteriskSCF::SessionCommunications::V1::SessionListenerPrx&, const PJSipManagerPtr& manager,
const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
const AsteriskSCF::System::Component::V1::ReplicaPtr& replica,
- bool ipv6, bool isUAC, const NATEndpointOptions& natOptions);
- bool isUAC, const SipEndpointConfig& config);
++ bool ipv6, bool isUAC, const SipEndpointConfig& config, const NATEndpointOptions& natOptions);
- SipSession(const Ice::ObjectAdapterPtr&, const SipEndpointPtr&, const std::string&, const Ice::Identity&,
- const Ice::Identity&, const AsteriskSCF::Replication::SipSessionManager::V1::RTPMediaSessionSeq&,
- const AsteriskSCF::Media::V1::StreamSourceSeq&, const AsteriskSCF::Media::V1::StreamSinkSeq&,
- PJSipManager *manager, const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
- const AsteriskSCF::System::Component::V1::ReplicaPtr& replica,
- bool isUAC, const SipEndpointConfig& config);
+ SipSession(const Ice::ObjectAdapterPtr&, const SipEndpointPtr&, const std::string&, const Ice::Identity&, const Ice::Identity&,
+ const Ice::Identity&, const AsteriskSCF::Replication::SipSessionManager::V1::RTPMediaSessionSeq&,
+ const AsteriskSCF::Media::V1::StreamSourceSeq&, const AsteriskSCF::Media::V1::StreamSinkSeq&,
+ const PJSipManagerPtr& manager, const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
+ const AsteriskSCF::System::Component::V1::ReplicaPtr& replica,
- bool isUAC, const NATEndpointOptions& natOptions);
++ bool isUAC, const SipEndpointConfig& config, const NATEndpointOptions& natOptions);
bool operator==(const SipSession &other) const;
@@@ -182,20 -174,27 +187,40 @@@
AsteriskSCF::SessionCommunications::V1::SessionCookies getCookies();
AsteriskSCF::SessionCommunications::V1::SessionCookieDict getAllCookies();
+ void getStreams_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_Session_getStreamsPtr&,
+ const Ice::Current&);
+
+ void setAndGetSessionController_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_Session_setAndGetSessionControllerPtr&,
+ const AsteriskSCF::SessionCommunications::V1::SessionControllerPrx&,
+ const Ice::Current&);
+
+ void removeSessionController_async(const AsteriskSCF::SessionCommunications::V1::AMD_Session_removeSessionControllerPtr&,
+ const AsteriskSCF::SessionCommunications::V1::SessionControllerPrx&,
+ const Ice::Current&);
+
/**
+ * TelephonySession overrides
+ */
+ void getSources_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSourcesPtr&,
+ const Ice::Current&);
+ /**
+ * Only called from within a queued operation
+ */
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourceSeq getSources();
+
+ void getSinks_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSinksPtr&,
+ const Ice::Current&);
+
+ /**
+ * Only called from within a queued operation
+ */
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq getSinks();
+
+ /**
* Implementation specific functions.
*/
void destroy();
diff --cc src/SipTelephonyEventSink.cpp
index 0000000,d7b320d..0e37c6c
mode 000000,100644..100644
--- a/src/SipTelephonyEventSink.cpp
+++ b/src/SipTelephonyEventSink.cpp
@@@ -1,0 -1,220 +1,220 @@@
+ /*
+ * Asterisk SCF -- An open-source communications framework.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk SCF project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE.txt file
+ * at the top of the source tree.
+ */
+
+ #include "SipTelephonyEventSink.h"
+
+ namespace AsteriskSCF
+ {
+
+ namespace SipSessionManager
+ {
+
+ using namespace AsteriskSCF::System::WorkQueue::V1;
+
+ class WriteTelephonyEvent : public SuspendableWork
+ {
+ public:
+ WriteTelephonyEvent(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_writePtr& cb,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventPtr& event,
+ pjsip_inv_session *inv)
+ : mCB(cb), mEvent(event), mInv(inv) { }
+
+ SuspendableWorkResult execute(const SuspendableWorkListenerPtr&)
+ {
+ AsteriskSCF::SessionCommunications::V1::BeginDTMFEventPtr beginDTMF;
+ AsteriskSCF::SessionCommunications::V1::EndDTMFEventPtr endDTMF;
+ AsteriskSCF::SessionCommunications::V1::FlashEventPtr flash;
+
+ if ((beginDTMF = AsteriskSCF::SessionCommunications::V1::BeginDTMFEventPtr::dynamicCast(mEvent)))
+ {
+ // XXX Since the only DTMF support we have is INFO, we won't get any beginDTMF indications. Sorry.
+ }
+ else if ((endDTMF = AsteriskSCF::SessionCommunications::V1::EndDTMFEventPtr::dynamicCast(mEvent)))
+ {
+ try
+ {
+ sendDTMFINFO(endDTMF);
+ }
+ catch (const AsteriskSCF::SessionCommunications::V1::TelephonyEventException& ex)
+ {
+ mCB->ice_exception(ex);
+ }
+ }
+ else if ((flash = AsteriskSCF::SessionCommunications::V1::FlashEventPtr::dynamicCast(mEvent)))
+ {
+ try
+ {
+ sendFlashINFO();
+ }
+ catch (const AsteriskSCF::SessionCommunications::V1::TelephonyEventException& ex)
+ {
+ mCB->ice_exception(ex);
+ }
+ }
+ mCB->ice_response();
+ return Complete;
+ }
+ private:
+
+ pjsip_tx_data *createINFORequest()
+ {
+ pjsip_method infoMethod;
+ pj_str_t infoStr;
+ pj_cstr(&infoStr, "INFO");
+ pjsip_method_init_np(&infoMethod, &infoStr);
+
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ status = pjsip_dlg_create_request(mInv->dlg, &infoMethod, -1, &tdata);
+
+ if (status != PJ_SUCCESS)
+ {
+ throw AsteriskSCF::SessionCommunications::V1::TelephonyEventException();
+ }
+
+ return tdata;
+ }
+
+ void fillINFOBody(pjsip_tx_data *tdata, char signal, int duration)
+ {
+ pj_str_t type;
+ pj_cstr(&type, "application");
+ pj_str_t subtype;
+ pj_cstr(&subtype, "dtmf-relay");
+ std::stringstream bodyText;
+ bodyText << "Signal= " << signal
+ << "\r\nDuration= " << duration << "\r\n\r\n";
+ pj_str_t bodyStr;
+ pj_cstr(&bodyStr, bodyText.str().c_str());
+
+ pjsip_msg_body *body = pjsip_msg_body_create(tdata->pool, &type, &subtype, &bodyStr);
+
+ if (!body)
+ {
+ throw AsteriskSCF::SessionCommunications::V1::TelephonyEventException();
+ }
+
+ tdata->msg->body = body;
+ }
+
+ void sendDTMFINFO(const AsteriskSCF::SessionCommunications::V1::EndDTMFEventPtr& endDTMF)
+ {
+ pjsip_tx_data *tdata = createINFORequest();
+
+ pj_str_t infoPackage;
+ pj_cstr(&infoPackage, "Info-Package");
+ pj_str_t infoPackageVal;
+ pj_cstr(&infoPackageVal, "dtmf");
+ pjsip_generic_string_hdr *infoPackageHdr = pjsip_generic_string_hdr_create(tdata->pool, &infoPackage, &infoPackageVal);
+
+ pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) infoPackageHdr);
+
- fillINFOBody(tdata, endDTMF->digit, endDTMF->duration);
++ fillINFOBody(tdata, endDTMF->signal, endDTMF->duration);
+
+ pjsip_dlg_send_request(mInv->dlg, tdata, -1, NULL);
+ }
+
+ void sendFlashINFO()
+ {
+ pjsip_tx_data *tdata = createINFORequest();
+ fillINFOBody(tdata, '!', 100);
+ pjsip_dlg_send_request(mInv->dlg, tdata, -1, NULL);
+ }
+
+ AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_writePtr mCB;
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventPtr mEvent;
+ pjsip_inv_session *mInv;
+ };
+
+
+ SipTelephonyEventSink::SipTelephonyEventSink(const SessionWorkPtr& sessionWork, pjsip_inv_session *inv)
+ : mSessionWork(sessionWork), mInv(inv) { }
+
+ void SipTelephonyEventSink::write_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_writePtr& cb,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventPtr& event,
+ const Ice::Current&)
+ {
+ mSessionWork->enqueueWork(new WriteTelephonyEvent(cb, event, mInv));
+ }
+
+ class SetTelephonyEventSource : public SuspendableWork
+ {
+ public:
+ SetTelephonyEventSource(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_setSourcePtr& cb,
+ const SipTelephonyEventSinkPtr& sink,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx& source)
- : mSink(sink), mSource(source) { }
++ : mCB(cb), mSink(sink), mSource(source) { }
+
+ SuspendableWorkResult execute(const SuspendableWorkListenerPtr&)
+ {
+ mSink->setSource(mSource);
+ mCB->ice_response();
+ return Complete;
+ }
+ private:
+ AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_setSourcePtr mCB;
+ SipTelephonyEventSinkPtr mSink;
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx mSource;
+ };
+
+ void SipTelephonyEventSink::setSource_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_setSourcePtr& cb,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx& source,
+ const Ice::Current&)
+ {
+ mSessionWork->enqueueWork(new SetTelephonyEventSource(cb, this, source));
+ }
+
+ void SipTelephonyEventSink::setSource(const AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx& source)
+ {
+ mSource = source;
+ }
+
+ class GetTelephonyEventSource : public SuspendableWork
+ {
+ public:
+ GetTelephonyEventSource(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_getSourcePtr& cb,
+ const SipTelephonyEventSinkPtr& sink)
- : mSink(sink) { }
++ : mCB(cb), mSink(sink) { }
+
+ SuspendableWorkResult execute(const SuspendableWorkListenerPtr&)
+ {
+ mCB->ice_response(mSink->getSource());
+ return Complete;
+ }
+ private:
+ AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_getSourcePtr mCB;
+ SipTelephonyEventSinkPtr mSink;
+ };
+
+ void SipTelephonyEventSink::getSource_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_getSourcePtr& cb,
+ const Ice::Current&)
+ {
+ mSessionWork->enqueueWork(new GetTelephonyEventSource(cb, this));
+ }
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx SipTelephonyEventSink::getSource()
+ {
+ return mSource;
+ }
+
+ };
+ };
diff --cc src/SipTelephonyEventSource.cpp
index 0000000,b7eab2c..b424dfa
mode 000000,100644..100644
--- a/src/SipTelephonyEventSource.cpp
+++ b/src/SipTelephonyEventSource.cpp
@@@ -1,0 -1,106 +1,108 @@@
+ /*
+ * Asterisk SCF -- An open-source communications framework.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk SCF project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE.txt file
+ * at the top of the source tree.
+ */
+
+ #include "SipTelephonyEventSource.h"
+ #include <AsteriskSCF/System/WorkQueue/WorkQueueIf.h>
+
+ namespace AsteriskSCF
+ {
+
+ namespace SipSessionManager
+ {
+
+ using namespace AsteriskSCF::System::WorkQueue::V1;
+
+ void SipTelephonyEventSource::distributeToSinks(const AsteriskSCF::SessionCommunications::V1::TelephonyEventPtr& event)
+ {
+ for (AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq::const_iterator iter = mSinks.begin();
+ iter != mSinks.end(); ++iter)
+ {
+ (*iter)->write(event);
+ }
+ }
+
+ SipTelephonyEventSource::SipTelephonyEventSource(const SessionWorkPtr& sessionWork)
+ : mSessionWork(sessionWork) { }
+
+ class AddSink : public SuspendableWork
+ {
+ public:
+ AddSink(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSource_addSinkPtr& cb,
+ const SipTelephonyEventSourcePtr& source,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkPrx& sink)
+ : mCB(cb), mSource(source), mSink(sink) { }
+
+ SuspendableWorkResult execute(const SuspendableWorkListenerPtr&)
+ {
+ mSource->addSink(mSink);
+ mCB->ice_response();
++ return Complete;
+ }
+ private:
... 4674 lines suppressed ...
--
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