[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
Thu May 19 11:58:37 CDT 2011


branch "master" has been updated
       via  8eb9a6461494bff1489e01b5541b569e7ecef3bc (commit)
       via  40f2cf7feed6d313e0f59a3188906960ea0eb84c (commit)
      from  958eccbe685be31e05b8651dd7bc3a72a3d11c99 (commit)

Summary of changes:
 config/Sip.config                  |    2 ++
 config/SipConfigurator.py          |    3 +++
 local-slice/SipConfigurationIf.ice |    5 +++++
 src/SipConfiguration.cpp           |   10 ++++++++++
 src/SipEndpoint.cpp                |    9 +++++++--
 src/SipEndpoint.h                  |    4 ++++
 src/SipSession.cpp                 |   33 +++++++++++++++++++++++----------
 src/SipSession.h                   |    5 +++--
 8 files changed, 57 insertions(+), 14 deletions(-)


- Log -----------------------------------------------------------------
commit 8eb9a6461494bff1489e01b5541b569e7ecef3bc
Merge: 958eccb 40f2cf7
Author: Joshua Colp <jcolp at digium.com>
Date:   Thu May 19 13:59:10 2011 -0300

    Merge branch 'rtp-ipv6'


commit 40f2cf7feed6d313e0f59a3188906960ea0eb84c
Author: Joshua Colp <jcolp at digium.com>
Date:   Tue May 17 14:02:32 2011 -0300

    Add support for RTP over IPv6.

diff --git a/config/Sip.config b/config/Sip.config
index 048a751..e1f3937 100644
--- a/config/Sip.config
+++ b/config/Sip.config
@@ -75,3 +75,5 @@ targetport=5060
 #sourceport=
 # What directions calls are permitted in. Valid options are inbound, outbound, and both.
 direction=both
+# Whether to use IPv6 for media transport or not
+rtpoveripv6=no
diff --git a/config/SipConfigurator.py b/config/SipConfigurator.py
index d04f2ac..4b820f2 100755
--- a/config/SipConfigurator.py
+++ b/config/SipConfigurator.py
@@ -111,6 +111,9 @@ class SipSectionVisitors(Configurator.SectionVisitors):
         transformer = AllowableCallDirectionTransformer(config)
         mapper.map('direction', AsteriskSCF.SIP.V1.SipAllowableCallDirectionItem(), 'callDirection', 'callDirection', transformer.get, None)
 
+        item = AsteriskSCF.SIP.V1.RTPMediaServiceItem()
+        mapper.map('rtpoveripv6', item, 'requireIPv6', 'mediaservice', config.getboolean, None)
+
         item = AsteriskSCF.SIP.V1.SipCryptoCertificateItem()
         mapper.map('certificateauthorityfile', item, 'certificateAuthority', 'cryptocert', config.get, None)
         mapper.map('certificatefile', item, 'certificate', 'cryptocert', config.get, None)
@@ -161,6 +164,10 @@ class SipSectionVisitors(Configurator.SectionVisitors):
         elif config.get(section, 'type') == 'endpoint':
             self.visit_endpoint(config, section)
 
+# In order to do service locator based lookup we need to pass in a params object
+serviceLocatorParams = AsteriskSCF.SIP.V1.SipConfigurationParams()
+serviceLocatorParams.category = AsteriskSCF.SIP.V1.ConfigurationDiscoveryCategory
+
 # Make a configurator application and let it run
-app = Configurator.ConfiguratorApp('Sip.config', SipSectionVisitors())
+app = Configurator.ConfiguratorApp('Sip.config', SipSectionVisitors(), None, serviceLocatorParams)
 sys.exit(app.main(sys.argv))
diff --git a/local-slice/SipConfigurationIf.ice b/local-slice/SipConfigurationIf.ice
index 58cae24..14f700a 100644
--- a/local-slice/SipConfigurationIf.ice
+++ b/local-slice/SipConfigurationIf.ice
@@ -241,6 +241,11 @@ module V1
        * Name of the RTP media service to use
        */
       string mediaServiceName;
+
+      /**
+       * Whether to choose an IPv6 RTP media service or not
+       */
+      bool requireIPv6 = false;
    };
 
    /**
diff --git a/src/SipConfiguration.cpp b/src/SipConfiguration.cpp
index bfd715d..1b40dad 100644
--- a/src/SipConfiguration.cpp
+++ b/src/SipConfiguration.cpp
@@ -194,6 +194,11 @@ class EndpointConfigHelper : public boost::enable_shared_from_this<EndpointConfi
             mUpdates.push_back(boost::bind(&EndpointConfigHelper::updateTarget, mConfig, target));
         };
 
+	void visitSipRTPMediaServiceItem(const SipRTPMediaServiceItemPtr& service)
+	{
+	    mUpdates.push_back(boost::bind(&EndpointConfigHelper::updateMediaService, mConfig, service));
+	};
+
     private:
 
         UpdateCommandList mUpdates;;
@@ -258,6 +263,11 @@ public:
         mEndpoint->setTargetAddress(target->host, target->port);
     }
 
+    void updateMediaService(const SipRTPMediaServiceItemPtr& service)
+    {
+	mEndpoint->setRTPOverIPv6(service->requireIPv6);
+    }
+
     void updated(const UpdateCommandList& updates)
     {
         //
diff --git a/src/SipEndpoint.cpp b/src/SipEndpoint.cpp
index e066cd4..c130685 100644
--- a/src/SipEndpoint.cpp
+++ b/src/SipEndpoint.cpp
@@ -149,6 +149,11 @@ void SipEndpoint::setCallDirection(enum Direction direction)
     mImplPriv->mConfig.sessionConfig.callDirection = direction;
 }
 
+void SipEndpoint::setRTPOverIPv6(bool enabled)
+{
+    mImplPriv->mConfig.sessionConfig.rtpOverIPv6 = enabled;
+}
+
 void SipEndpoint::setConfiguration(const Ice::PropertyDict& props)
 {
     setTransportConfiguration(props);
@@ -265,7 +270,7 @@ AsteriskSCF::SessionCommunications::V1::SessionPrx SipEndpoint::createSession(co
     }
 
     SipSessionPtr session = new SipSession(mImplPriv->mAdapter, this, destination, listener, mImplPriv->mManager,
-            mImplPriv->mServiceLocator, mImplPriv->mReplica);
+	mImplPriv->mServiceLocator, mImplPriv->mReplica, mImplPriv->mConfig.sessionConfig.rtpOverIPv6);
     mImplPriv->mSessions.push_back(session);
     return session->getSessionProxy();
 }
@@ -273,7 +278,7 @@ AsteriskSCF::SessionCommunications::V1::SessionPrx SipEndpoint::createSession(co
 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->mServiceLocator, mImplPriv->mReplica, mImplPriv->mConfig.sessionConfig.rtpOverIPv6);
     mImplPriv->mSessions.push_back(session);
     return session;
 }
diff --git a/src/SipEndpoint.h b/src/SipEndpoint.h
index 1eb9680..2f5e011 100644
--- a/src/SipEndpoint.h
+++ b/src/SipEndpoint.h
@@ -138,6 +138,9 @@ public:
     // The source IP address and port to use
     // when contacting this endpoint.
     std::string sourceAddress;
+    // Whether we are using IPv6 for media transport
+    // or not.
+    bool rtpOverIPv6;
 };
 
 /**
@@ -254,6 +257,7 @@ public:
     void setSourceAddress(std::string, int);
     void setTargetAddress(std::string, int);
     void setCallDirection(enum Direction);
+    void setRTPOverIPv6(bool);
 
 private:
     /**
diff --git a/src/SipSession.cpp b/src/SipSession.cpp
index 2b36d1f..6f06e3f 100644
--- a/src/SipSession.cpp
+++ b/src/SipSession.cpp
@@ -173,7 +173,7 @@ public:
 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)
+    const AsteriskSCF::System::Component::V1::ReplicaPtr& replica, bool ipv6)
     : mImplPriv(new SipSessionPriv(adapter, endpoint, destination, manager, serviceLocator, replica))
 {
     if (listener != 0)
@@ -190,7 +190,7 @@ SipSession::SipSession(const Ice::ObjectAdapterPtr& adapter, const SipEndpointPt
 
     // Get an RTP session capable of handling the formats we are going to offer
     AsteriskSCF::Media::V1::FormatSeq formats;
-    requestRTPSessions(formats);
+    requestRTPSessions(formats, ipv6);
 }
 
 /**
@@ -560,22 +560,33 @@ pjmedia_sdp_session *SipSession::createSDPOffer()
     pj_gettimeofday(&tv);
     sdp->origin.version = sdp->origin.id = (pj_uint32_t) (tv.sec + 2208988800UL);
     pj_strdup2(mImplPriv->mDialog->pool, &sdp->origin.net_type, "IN");
-    pj_strdup2(mImplPriv->mDialog->pool, &sdp->origin.addr_type, "IP4");
+
+    // Right now we only support a single stream so go and get it
+    AsteriskSCF::Media::RTP::V1::StreamSourceRTPPrx stream =
+        AsteriskSCF::Media::RTP::V1::StreamSourceRTPPrx::uncheckedCast(mImplPriv->mSources.front());
+
+    std::string address = stream->getLocalAddress();
+
+    if (address.find(":") != std::string::npos)
+    {
+	pj_strdup2(mImplPriv->mDialog->pool, &sdp->origin.addr_type, "IP6");
+    }
+    else
+    {
+	pj_strdup2(mImplPriv->mDialog->pool, &sdp->origin.addr_type, "IP4");
+    }
+
     sdp->origin.addr = *pj_gethostname();
     sdp->name = sdp->origin.user;
     sdp->time.start = 0;
     sdp->time.stop = 0;
     sdp->attr_count = 0;
 
-    // Right now we only support a single stream so go and get it
-    AsteriskSCF::Media::RTP::V1::StreamSourceRTPPrx stream =
-        AsteriskSCF::Media::RTP::V1::StreamSourceRTPPrx::uncheckedCast(mImplPriv->mSources.front());
-
     // Add connection details at the session level since we currently only support one media stream.
     sdp->conn = static_cast<pjmedia_sdp_conn*>(pj_pool_zalloc(mImplPriv->mDialog->pool, sizeof(pjmedia_sdp_conn)));
     sdp->conn->net_type = sdp->origin.net_type;
     sdp->conn->addr_type = sdp->origin.addr_type;
-    pj_strdup2(mImplPriv->mDialog->pool, &sdp->conn->addr, stream->getLocalAddress().c_str());
+    pj_strdup2(mImplPriv->mDialog->pool, &sdp->conn->addr, address.c_str());
 
     // Add a single media stream
     sdp->media_count = 1;
@@ -615,16 +626,18 @@ pjmedia_sdp_session *SipSession::createSDPOffer()
 /**
  * Internal function called to request needed RTP sessions.
  */
-void SipSession::requestRTPSessions(AsteriskSCF::Media::V1::FormatSeq& formats)
+void SipSession::requestRTPSessions(AsteriskSCF::Media::V1::FormatSeq& formats, bool ipv6)
 {
     // TODO: This needs to be improved for multiple streams
     AsteriskSCF::Media::RTP::V1::RTPServiceLocatorParamsPtr params =
         new AsteriskSCF::Media::RTP::V1::RTPServiceLocatorParams();
     params->category = "rtp";
+    params->formats = formats;
+    params->ipv6 = ipv6;
 
     AsteriskSCF::Media::RTP::V1::RTPMediaServicePrx factory =
         AsteriskSCF::Media::RTP::V1::RTPMediaServicePrx::uncheckedCast(mImplPriv->mServiceLocator->locate(params));
-    AsteriskSCF::Media::RTP::V1::RTPSessionPrx session = factory->allocate(formats);
+    AsteriskSCF::Media::RTP::V1::RTPSessionPrx session = factory->allocate(params);
     mImplPriv->mRTPSessions.push_back(session);
 
     // Create a local copy of the sources, this won't get changed by the RTP session so it's all good
diff --git a/src/SipSession.h b/src/SipSession.h
index 643e70e..e18c480 100644
--- a/src/SipSession.h
+++ b/src/SipSession.h
@@ -62,7 +62,8 @@ public:
     SipSession(const Ice::ObjectAdapterPtr&, const SipEndpointPtr&, const std::string&,
         const AsteriskSCF::SessionCommunications::V1::SessionListenerPrx&, PJSipManager *manager,
         const AsteriskSCF::Core::Discovery::V1::ServiceLocatorPrx& serviceLocator,
-        const AsteriskSCF::System::Component::V1::ReplicaPtr& replica);
+        const AsteriskSCF::System::Component::V1::ReplicaPtr& replica,
+	bool ipv6);
 
     SipSession(const Ice::ObjectAdapterPtr&, const SipEndpointPtr&, const std::string&, const Ice::Identity&,
         const Ice::Identity&, const AsteriskSCF::Media::V1::SessionPrx&,
@@ -133,7 +134,7 @@ public:
 
     AsteriskSCF::Media::V1::SessionPrx getHiddenMediaSession();
 private:
-    void requestRTPSessions(AsteriskSCF::Media::V1::FormatSeq& formats);
+    void requestRTPSessions(AsteriskSCF::Media::V1::FormatSeq& formats, bool ipv6);
 
     /**
      * Private implementation details.

-----------------------------------------------------------------------


-- 
asterisk-scf/release/sip.git



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