[asterisk-scf-commits] asterisk-scf/integration/sip.git branch "srtp-support" updated.

[Commits to the Asterisk SCF project code repositories]

branch "srtp-support" has been updated
       via  cfec4e219a70134f607888d309025fa219f1a8e9 (commit)
       via  5c058d2ef2e1c96e9eb538ed6cd8c8ac1614d967 (commit)
       via  5846372d877d813e35164d59177c7dffd3324d65 (commit)
      from  726751650fb57aea3497979a4e3a43d97ca99e3a (commit)

Summary of changes:
 src/STUNTransport.cpp |    1 +
 src/SipSession.cpp    |   92 ++++++++++++++++++++++++++----------------------
 2 files changed, 51 insertions(+), 42 deletions(-)
 mode change 100644 => 100755 src/SipSession.cpp


- Log -----------------------------------------------------------------
commit cfec4e219a70134f607888d309025fa219f1a8e9
Merge: 5c058d2 5846372
Author: Brent Eagles <beagles at digium.com>
Date:   Tue Jul 12 10:09:53 2011 -0230

    Update srtp-support with latest commits pushed to nat-support.
    
    Acquired and resolved some conflicts:
    	src/SipSession.cpp

diff --cc src/SipSession.cpp
index d9bba00,43f12c7..8f613ad
mode 100644,100755..100755
--- a/src/SipSession.cpp
+++ b/src/SipSession.cpp
@@@ -1340,61 -1301,51 +1340,69 @@@ void SipSession::requestRTPSessions(Ast
      params->category = "rtp";
      params->formats = formats;
      params->ipv6 = ipv6;
 +    params->srtpCapable = (config.srtpConfig.supportOptions != SipEndpointMediaSRTPConfig::Disabled);
  
-     AsteriskSCF::Media::RTP::V1::RTPMediaServicePrx factory =
-         AsteriskSCF::Media::RTP::V1::RTPMediaServicePrx::uncheckedCast(mImplPriv->mServiceLocator->locate(params));
-     if (!factory)
+     try
      {
-         assert("Location request has returned a nil proxy, this should throw!" == 0);
-         //
-         // TODO: throw!
-         //
-     }
-     AsteriskSCF::Media::RTP::V1::RTPSessionPrx session = factory->allocate(params);
+         AsteriskSCF::Media::RTP::V1::RTPMediaServicePrx factory =
+             AsteriskSCF::Media::RTP::V1::RTPMediaServicePrx::uncheckedCast(mImplPriv->mServiceLocator->locate(params));
+         if (!factory)
+         {
+             assert("Location request has returned a nil proxy, this should throw!" == 0);
+             //
+             // TODO: throw!
+             //
+         }
+         AsteriskSCF::Media::RTP::V1::RTPSessionPrx session = factory->allocate(params);
  
-     if (!session)
-     {
-         assert("Allocation of rtp streams failed! This should've thrown an exception!" == 0);
-         //
-         // TODO Throw!
-         //
-     }
-     mImplPriv->mRTPSessions.push_back(session);
+         if (!session)
+         {
+             assert("Allocation of rtp streams failed! This should've thrown an exception!" == 0);
+             //
+             // TODO Throw!
+             //
+         }
+         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
-     mImplPriv->mSources = session->getSources();
+         // Create a local copy of the sources, this won't get changed by the RTP session so it's all good
+         mImplPriv->mSources = session->getSources();
  
-     // Create a local copy of the sinks, this won't get changed by the RTP session so it's all good
-     mImplPriv->mSinks = session->getSinks();
+         // Create a local copy of the sinks, this won't get changed by the RTP session so it's all good
+         mImplPriv->mSinks = session->getSinks();
  
-     // For testing push a static payload mapping for ULAW into the RTP session
-     AsteriskSCF::Media::RTP::V1::PayloadMap payloads;
-     AsteriskSCF::Media::V1::AudioFormatPtr format = new AsteriskSCF::Media::V1::AudioFormat();
-     format->name = "ulaw";
-     format->sampleRate = 8000;
-     format->frameSize = 20;
-     format->maximumFrameSize = 20;
-     format->minimumFrameSize = 20;
-     payloads.insert(std::make_pair(0, format));
+         // For testing push a static payload mapping for ULAW into the RTP session
+         AsteriskSCF::Media::RTP::V1::PayloadMap payloads;
+         AsteriskSCF::Media::V1::AudioFormatPtr format = new AsteriskSCF::Media::V1::AudioFormat();
+         format->name = "ulaw";
+         format->sampleRate = 8000;
+         format->frameSize = 20;
+         format->maximumFrameSize = 20;
+         format->minimumFrameSize = 20;
+         payloads.insert(std::make_pair(0, format));
  
-     session->associatePayloads(payloads);
+         session->associatePayloads(payloads);
 +
-     if (params->srtpCapable && !config.srtpConfig.cryptoKeys.empty())
-     {
-         SRTPSessionPrx srtpPrx = SRTPSessionPrx::checkedCast(session);
-         if (!srtpPrx && config.srtpConfig.supportOptions == SipEndpointMediaSRTPConfig::Required)
++        if (params->srtpCapable && !config.srtpConfig.cryptoKeys.empty())
 +        {
++            SRTPSessionPrx srtpPrx = SRTPSessionPrx::checkedCast(session);
++            if (!srtpPrx && config.srtpConfig.supportOptions == SipEndpointMediaSRTPConfig::Required)
++            {
++                //
++                // TODO throw.
++                //
++                assert("Cannot access SRTP functionality in the RTPSession." == 0);
++            }
 +            //
-             // TODO throw.
++            // This effectively treats this as if it is required SRTP.
 +            //
-             assert("Cannot access SRTP functionality in the RTPSession." == 0);
++            srtpPrx->setOptions(config.srtpConfig.cryptoKeys[0].suite, config.srtpConfig.cryptoKeys[0].cryptoKey,
++              config.srtpConfig.enableAuthentication, config.srtpConfig.enableEncryption);
 +        }
-         //
-         // This effectively treats this as if it is required SRTP.
-         //
-         srtpPrx->setOptions(config.srtpConfig.cryptoKeys[0].suite, config.srtpConfig.cryptoKeys[0].cryptoKey,
-                 config.srtpConfig.enableAuthentication, config.srtpConfig.enableEncryption);
+     }
+     catch (const Ice::Exception& ex)
+     {
+         lg(Debug) << "Exception caught while trying to allocate RTP session. " << ex.what();
+         throw;
      }
  }
  

commit 5c058d2ef2e1c96e9eb538ed6cd8c8ac1614d967
Author: Brent Eagles <beagles at digium.com>
Date:   Tue Jul 12 10:02:08 2011 -0230

    Fix missing return value.

diff --git a/src/STUNTransport.cpp b/src/STUNTransport.cpp
index 1b509b4..076ee72 100644
--- a/src/STUNTransport.cpp
+++ b/src/STUNTransport.cpp
@@ -120,6 +120,7 @@ private:
         {
             data->op_key.tdata = 0;
         }
+        return PJ_SUCCESS;
     }
 
     static pj_status_t shutdownImplCB(pjsip_transport*)

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


-- 
asterisk-scf/integration/sip.git