[asterisk-scf-commits] asterisk-scf/integration/media_rtp_pjmedia.git branch "rtcp" updated.

[Commits to the Asterisk SCF project code repositories]

branch "rtcp" has been updated
       via  2e2923e09eb4045a15578884de929a982868e531 (commit)
      from  3eb956212440c9b0a32d9c5639c8c18ec186ed5e (commit)

Summary of changes:
 src/RTPSession.cpp |    6 ++----
 src/RTPSource.cpp  |    5 -----
 2 files changed, 2 insertions(+), 9 deletions(-)


- Log -----------------------------------------------------------------
commit 2e2923e09eb4045a15578884de929a982868e531
Author: Joshua Colp <jcolp at digium.com>
Date:   Tue Jul 12 09:33:09 2011 -0300

    Fix a bug where RTP passing before RTCP was finalized would cause a crash due to dividing by zero. This is now fixed by initializing the RTCP session to a sane default state when created.

diff --git a/src/RTPSession.cpp b/src/RTPSession.cpp
index f645c16..6ad9b67 100644
--- a/src/RTPSession.cpp
+++ b/src/RTPSession.cpp
@@ -153,10 +153,8 @@ public:
         mSessionStateItem(new RtpSessionStateItem()),
         mReplicaService(replicaService), mStateReplicator(stateReplicator)
     {
-        pjmedia_rtcp_session_setting defaults;
-
-        pjmedia_rtcp_session_setting_default(&defaults);
-        pjmedia_rtcp_init2(&mRtcpSession, &defaults);
+        // This will need to be adjusted elsewhere as the codec used changes, along with SSRC
+        pjmedia_rtcp_init(&mRtcpSession, NULL, 8000, 160, 0);
     }
 
     ~RTPSessionImplPriv();
diff --git a/src/RTPSource.cpp b/src/RTPSource.cpp
index 6757d68..5e2526b 100644
--- a/src/RTPSource.cpp
+++ b/src/RTPSource.cpp
@@ -501,11 +501,6 @@ void StreamSourceRTPImpl::setRemoteRtcpDetails(const std::string& address, Ice::
         throw InvalidAddress();
     }
 
-    // Since the RTCP is going to a new destination initialize the session to a fresh state
-
-    // TODO - The clock rate and samples per frame will need to change, along with SSRC
-    pjmedia_rtcp_init(mImpl->mSession->getRtcpSession(), NULL, 8000, 160, 0);
-
     // If RTCP is not already being sent start sending it
     if (!mImpl->mTimer && (mImpl->mTimer = new IceUtil::Timer()))
     {

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-- 
asterisk-scf/integration/media_rtp_pjmedia.git