[asterisk-scf-commits] asterisk-scf/release/slice.git branch "master" updated.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Thu May 19 11:34:20 CDT 2011
branch "master" has been updated
via 941c14fc79f8c2e47957b2db994a9412734a49d0 (commit)
from c06dbbefb20ab2f3bb3b2fbdf1eb714d304af928 (commit)
Summary of changes:
AsteriskSCF/Media/RTP/MediaRTPIf.ice | 22 ++++++++++++++++++----
1 files changed, 18 insertions(+), 4 deletions(-)
- Log -----------------------------------------------------------------
commit 941c14fc79f8c2e47957b2db994a9412734a49d0
Author: Joshua Colp <jcolp at digium.com>
Date: Sun May 8 11:57:15 2011 -0300
Add support for IPv6.
diff --git a/AsteriskSCF/Media/RTP/MediaRTPIf.ice b/AsteriskSCF/Media/RTP/MediaRTPIf.ice
index b0cee7f..8687f30 100644
--- a/AsteriskSCF/Media/RTP/MediaRTPIf.ice
+++ b/AsteriskSCF/Media/RTP/MediaRTPIf.ice
@@ -43,6 +43,11 @@ module V1
* A sequence of formats that the RTP media service is expected to transport.
*/
AsteriskSCF::Media::V1::FormatSeq formats;
+
+ /**
+ * Whether IPv6 is to be used or not.
+ */
+ bool ipv6 = false;
};
/**
@@ -66,6 +71,13 @@ module V1
};
/**
+ * Exception thrown when an invalid address is passed in.
+ */
+ exception InvalidAddress
+ {
+ };
+
+ /**
* Interface to an RTP stream sink.
*/
interface StreamSinkRTP extends AsteriskSCF::Media::V1::StreamSink
@@ -73,11 +85,13 @@ module V1
/**
* Method which changes the IP address and port that media will be sent to.
*
- * @param address A string representation of the IP address.
+ * @param address A string representation of the address.
*
* @param port An integer containing the port.
+ *
+ * @throws InvalidAddress when the address passed in is invalid.
*/
- void setRemoteDetails(string address, int port);
+ void setRemoteDetails(string address, int port) throws InvalidAddress;
/**
* Method which retrieves the remote IP address.
@@ -146,11 +160,11 @@ module V1
/**
* Method which creates a new RTP session and returns a proxy to it.
*
- * @param formats The media formats the session is expected to carry.
+ * @param params Parameters to configure the RTP session with.
*
* @return RTPSession* A proxy to the new RTP session.
*/
- RTPSession* allocate(AsteriskSCF::Media::V1::FormatSeq formats);
+ RTPSession* allocate(RTPServiceLocatorParams params);
};
}; /* end module V1 */
-----------------------------------------------------------------------
--
asterisk-scf/release/slice.git
More information about the asterisk-scf-commits
mailing list