diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-06 16:09:33 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-06 16:09:33 +0000 |
commit | 4506e03f3d24f4bd4f6b17803031ede1e353de28 (patch) | |
tree | b849a5c247c05d8e052b854defc2526e0f341a64 /include | |
parent | 77abfdfd00db4460aeb6e9c3b5065406486ca4f4 (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.h | 12 | ||||
-rw-r--r-- | include/asterisk/translate.h | 3 |
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" { |