[asterisk-scf-commits] asterisk-scf/integration/slice.git branch "nat-traversal" updated.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Mon Jun 20 07:59:29 CDT 2011
branch "nat-traversal" has been updated
via adb9cffd39577fe4c31afaf06aafa3b7949e8802 (commit)
from 20f58b092c28497dad52c9840a7d69130ac154c2 (commit)
Summary of changes:
AsteriskSCF/Media/RTP/MediaRTPIf.ice | 33 ++++++++++++++++++++++++++++++++-
1 files changed, 32 insertions(+), 1 deletions(-)
- Log -----------------------------------------------------------------
commit adb9cffd39577fe4c31afaf06aafa3b7949e8802
Author: Brent Eagles <beagles at digium.com>
Date: Mon Jun 20 10:27:05 2011 -0230
Added a derived class for ICE enabled RTP transports. I vacillated on whether
the changes belonged in Media RTP and finally decided to leave it in the
MediaRTP slice because its more of an additional capability than a different
type of RTP stream.
diff --git a/AsteriskSCF/Media/RTP/MediaRTPIf.ice b/AsteriskSCF/Media/RTP/MediaRTPIf.ice
index 8687f30..d59ba3e 100644
--- a/AsteriskSCF/Media/RTP/MediaRTPIf.ice
+++ b/AsteriskSCF/Media/RTP/MediaRTPIf.ice
@@ -50,6 +50,25 @@ module V1
bool ipv6 = false;
};
+ /**
+ * Extend discovery class to enable RTP over ICE negotiated media flows.
+ */
+ unsliceable class RTPOverICEServiceLocatorParams extends RTPServiceLocatorParams
+ {
+ /**
+ * Enable ICE negotiated RTP media flows. We set a default of true mostly because
+ * we assume if you looking for a component that understands this type then
+ * it probably has this feature enabled.
+ */
+ bool enableRTPOverICE = true;
+
+ /**
+ * Enable TURN server access/usage if configured. NOTE: Has no effect if enableRTPOverICE
+ * is NOT enabled.
+ */
+ bool enableTURN = true;
+ };
+
/**
* Interface to an RTP stream source.
*/
@@ -153,6 +172,15 @@ module V1
};
/**
+ * This exception will be thrown by the media service if the session cannot be allocated for some
+ * reason. It shall include a descriptive message as to why the request failed.
+ */
+ exception SessionAllocationFailure
+ {
+ string message;
+ };
+
+ /**
* Interface to an RTP media service.
*/
interface RTPMediaService
@@ -163,8 +191,11 @@ module V1
* @param params Parameters to configure the RTP session with.
*
* @return RTPSession* A proxy to the new RTP session.
+ *
+ * @throws SessionAllocationFailure if the media service is unable to allocate a session
+ * to match the provided parameters.
*/
- RTPSession* allocate(RTPServiceLocatorParams params);
+ RTPSession* allocate(RTPServiceLocatorParams params) throws SessionAllocationFailure;
};
}; /* end module V1 */
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asterisk-scf/integration/slice.git
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