From 7c884d76ea44ffda6d0d49b6da121a0953cc6429 Mon Sep 17 00:00:00 2001
From: Russell Bryant <russell@russellbryant.com>
Date: Tue, 27 Mar 2007 16:25:02 +0000
Subject: [PATCH] Merged revisions 59256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines
Convert the RTPQOS function to just be additional parameter of the CHANNEL
function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 32 ++++++++------------------------
funcs/func_channel.c | 43 +++++++++++++++++++++++++++++++------------
2 files changed, 39 insertions(+), 36 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 28ea517f5b..d9639fb1d2 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1428,6 +1428,7 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data);
static int sip_addheader(struct ast_channel *chan, void *data);
static int sip_do_reload(enum channelreloadreason reason);
static int sip_reload(int fd, int argc, char *argv[]);
+static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
/*--- Debugging
Functions for enabling debug per IP or fully, or enabling history logging for
@@ -1593,6 +1594,7 @@ static const struct ast_channel_tech sip_tech = {
.bridge = ast_rtp_bridge,
.early_bridge = ast_rtp_early_bridge,
.send_text = sip_sendtext,
+ .func_channel_read = acf_channel_read,
};
/*! \brief This version of the sip channel tech has no send_digit_begin
@@ -14800,12 +14802,13 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
}
}
-static int acf_rtpqos_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
+static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
{
struct ast_rtp_quality qos;
struct sip_pvt *p = chan->tech_pvt;
char *all = "", *parse = ast_strdupa(preparse);
AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(param);
AST_APP_ARG(type);
AST_APP_ARG(field);
);
@@ -14814,8 +14817,12 @@ static int acf_rtpqos_read(struct ast_channel *chan, const char *funcname, char
/* Sanity check */
if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) {
ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
+ return 0;
}
+ if (!strcasecmp(args.param, "rtpqos"))
+ return 0;
+
memset(buf, 0, buflen);
memset(&qos, 0, sizeof(qos));
@@ -18135,27 +18142,6 @@ static struct ast_cli_entry cli_sip[] = {
sip_reload_usage },
};
-struct ast_custom_function acf_rtpqos = {
- .name = "RTPQOS",
- .synopsis = "Retrieve statistics about an RTP stream",
- .desc =
-"The following statistics may be retrieved:\n"
-" local_ssrc - Local SSRC (stream ID)\n"
-" local_lostpackets - Local lost packets\n"
-" local_jitter - Local calculated jitter\n"
-" local_count - Number of received packets\n"
-" remote_ssrc - Remote SSRC (stream ID)\n"
-" remote_lostpackets - Remote lost packets\n"
-" remote_jitter - Remote reported jitter\n"
-" remote_count - Number of transmitted packets\n"
-" rtt - Round trip time\n"
-" all - All statistics (in a form suited to logging, but not for parsing)\n"
-"\n"
-"Type may be specified as \"audio\", \"video\", or \"text\".\n",
- .syntax = "RTPQOS(<type>|<field>)",
- .read = acf_rtpqos_read,
-};
-
/*! \brief PBX load module - initialization */
static int load_module(void)
{
@@ -18205,7 +18191,6 @@ static int load_module(void)
ast_custom_function_register(&sippeer_function);
ast_custom_function_register(&sipchaninfo_function);
ast_custom_function_register(&checksipdomain_function);
- ast_custom_function_register(&acf_rtpqos);
/* Register manager commands */
ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers,
@@ -18235,7 +18220,6 @@ static int unload_module(void)
ast_custom_function_unregister(&sippeer_function);
ast_custom_function_unregister(&sip_header_function);
ast_custom_function_unregister(&checksipdomain_function);
- ast_custom_function_unregister(&acf_rtpqos);
/* Unregister dial plan applications */
ast_unregister_application(app_dtmfmode);
diff --git a/funcs/func_channel.c b/funcs/func_channel.c
index 0820baa4f1..4a6a9e221d 100644
--- a/funcs/func_channel.c
+++ b/funcs/func_channel.c
@@ -150,18 +150,37 @@ static struct ast_custom_function channel_function = {
.syntax = "CHANNEL(item)",
.desc = "Gets/set various pieces of information about the channel.\n"
"Standard items (provided by all channel technologies) are:\n"
- "R/O audioreadformat format currently being read\n"
- "R/O audionativeformat format used natively for audio\n"
- "R/O audiowriteformat format currently being written\n"
- "R/W callgroup call groups for call pickup\n"
- "R/O channeltype technology used for channel\n"
- "R/W language language for sounds played\n"
- "R/W musicclass class (from musiconhold.conf) for hold music\n"
- "R/W rxgain set rxgain level on channel drivers that support it\n"
- "R/O state state for channel\n"
- "R/W tonezone zone for indications played\n"
- "R/W txgain set txgain level on channel drivers that support it\n"
- "R/O videonativeformat format used natively for video\n"
+ "R/O audioreadformat format currently being read\n"
+ "R/O audionativeformat format used natively for audio\n"
+ "R/O audiowriteformat format currently being written\n"
+ "R/W callgroup call groups for call pickup\n"
+ "R/O channeltype technology used for channel\n"
+ "R/W language language for sounds played\n"
+ "R/W musicclass class (from musiconhold.conf) for hold music\n"
+ "R/W rxgain set rxgain level on channel drivers that support it\n"
+ "R/O state state for channel\n"
+ "R/W tonezone zone for indications played\n"
+ "R/W txgain set txgain level on channel drivers that support it\n"
+ "R/O videonativeformat format used natively for video\n"
+ "\n"
+ "chan_sip provides the following additional options:\n"
+ "R/O rtpqos Get QOS information about the RTP stream\n"
+ " This option takes two additional arguments:\n"
+ " Argument 1:\n"
+ " audio Get data about the audio stream\n"
+ " video Get data about the video stream\n"
+ " text Get data about the text stream\n"
+ " Argument 2:\n"
+ " local_ssrc Local SSRC (stream ID)\n"
+ " local_lostpackets Local lost packets\n"
+ " local_jitter Local calculated jitter\n"
+ " local_count Number of received packets\n"
+ " remote_ssrc Remote SSRC (stream ID)\n"
+ " remote_lostpackets Remote lost packets\n"
+ " remote_jitter Remote reported jitter\n"
+ " remote_count Number of transmitted packets\n"
+ " rtt Round trip time\n"
+ " all All statistics (in a form suited to logging, but not for parsing)\n"
"\n"
"Additional items may be available from the channel driver providing\n"
"the channel; see its documentation for details.\n"
--
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