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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-06-06 16:09:33 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-06-06 16:09:33 +0000
commit4506e03f3d24f4bd4f6b17803031ede1e353de28 (patch)
treeb849a5c247c05d8e052b854defc2526e0f341a64 /include
parent77abfdfd00db4460aeb6e9c3b5065406486ca4f4 (diff)
Merge of the "sdpcleanup" branch. Thanks to John Martin for a lot of tests
and some patches (all disclaimed). - Don't change RTP properties if we reject a re-INVITE - Don't add video to an outbound channel if there's no video on the inbound channel - Don't include video in the "preferred codec" list for codec selection - Clean up and document code that parses and adds SDP attachments Since we do not transcode video, we can't handle video the same way as audio. This is a bug fix patch. In future releases, we need to work on a solution for video negotiation, not codecs but formats and framerates instead. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32597 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/rtp.h12
-rw-r--r--include/asterisk/translate.h3
2 files changed, 9 insertions, 6 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 521819318..4410d7fb6 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -40,13 +40,13 @@ extern "C" {
/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
/*! DTMF (RFC2833) */
-#define AST_RTP_DTMF (1 << 0)
+#define AST_RTP_DTMF (1 << 0)
/*! 'Comfort Noise' (RFC3389) */
-#define AST_RTP_CN (1 << 1)
+#define AST_RTP_CN (1 << 1)
/*! DTMF (Cisco Proprietary) */
-#define AST_RTP_CISCO_DTMF (1 << 2)
+#define AST_RTP_CISCO_DTMF (1 << 2)
/*! Maximum RTP-specific code */
-#define AST_RTP_MAX AST_RTP_CISCO_DTMF
+#define AST_RTP_MAX AST_RTP_CISCO_DTMF
#define MAX_RTP_PT 256
@@ -62,6 +62,9 @@ struct ast_rtp_protocol {
AST_LIST_ENTRY(ast_rtp_protocol) list;
};
+
+#define FLAG_3389_WARNING (1 << 0)
+
typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
@@ -71,7 +74,6 @@ typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *
* RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]"
*
*/
-
/*! \brief The value of each payload format mapping: */
struct rtpPayloadType {
int isAstFormat; /*!< whether the following code is an AST_FORMAT */
diff --git a/include/asterisk/translate.h b/include/asterisk/translate.h
index 182330ace..29c3fd56c 100644
--- a/include/asterisk/translate.h
+++ b/include/asterisk/translate.h
@@ -23,7 +23,8 @@
#ifndef _ASTERISK_TRANSLATE_H
#define _ASTERISK_TRANSLATE_H
-#define MAX_FORMAT 32
+//#define MAX_FORMAT 15 /* Do not include video here */
+#define MAX_FORMAT 32 /* Do include video here */
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {