diff options
| author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-03-26 17:51:27 +0000 |
|---|---|---|
| committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-03-26 17:51:27 +0000 |
| commit | c04c322b066ac7fab8f0d4bf3a6205ba972fd22c (patch) | |
| tree | bb9a4f8a98d5844711b2d23378a49d0921f3f588 /include | |
| parent | b00932def0c2da02fba1b96b7e47723a99b779a8 (diff) | |
Merged revisions 59207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59208 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
| -rw-r--r-- | include/asterisk/rtp.h | 15 |
1 files changed, 13 insertions, 2 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index 5eea063db..47cc26212 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -84,6 +84,17 @@ struct ast_rtp_protocol { AST_LIST_ENTRY(ast_rtp_protocol) list; }; +struct ast_rtp_quality { + unsigned int local_ssrc; /* Our SSRC */ + unsigned int local_lostpackets; /* Our lost packets */ + double local_jitter; /* Our calculated jitter */ + unsigned int local_count; /* Number of received packets */ + unsigned int remote_ssrc; /* Their SSRC */ + unsigned int remote_lostpackets; /* Their lost packets */ + double remote_jitter; /* Their reported jitter */ + unsigned int remote_count; /* Number of transmitted packets */ + double rtt; /* Round trip time */ +}; /*! RTP callback structure */ typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data); @@ -219,12 +230,12 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i having to send a re-invite later */ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1); - +/*! \brief Return RTCP quality string */ +char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual); /*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */ int ast_rtcp_send_h261fur(void *data); -char *ast_rtp_get_quality(struct ast_rtp *rtp); /*! \brief Return RTCP quality string */ void ast_rtp_init(void); /*! Initialize RTP subsystem */ int ast_rtp_reload(void); /*! reload rtp configuration */ void ast_rtp_new_init(struct ast_rtp *rtp); |