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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-03-27 16:20:53 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-03-27 16:20:53 +0000
commit319bebfe27345d8f322569678d52e066e2208d66 (patch)
treeecdc297bf137319ba5b68328f273d7530c82c768
parentf601b1759e6918f6dd9ba2c5489c4e3cde4d1474 (diff)
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: http://svn.digium.com/svn/asterisk/branches/1.4@59256 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c32
-rw-r--r--funcs/func_channel.c42
2 files changed, 38 insertions, 36 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 086a302c7..b61052966 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1384,6 +1384,7 @@ 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, char *funcname, char *preparse, char *buf, size_t buflen);
/*--- Debugging
Functions for enabling debug per IP or fully, or enabling history logging for
@@ -1548,6 +1549,7 @@ static const struct ast_channel_tech sip_tech = {
.send_digit_end = sip_senddigit_end,
.bridge = ast_rtp_bridge,
.send_text = sip_sendtext,
+ .func_channel_read = acf_channel_read,
};
/*! \brief This version of the sip channel tech has no send_digit_begin
@@ -14127,12 +14129,13 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
}
}
-static int acf_rtpqos_read(struct ast_channel *chan, char *funcname, char *preparse, char *buf, size_t buflen)
+static int acf_channel_read(struct ast_channel *chan, 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);
);
@@ -14141,8 +14144,12 @@ static int acf_rtpqos_read(struct ast_channel *chan, char *funcname, char *prepa
/* 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));
@@ -17373,27 +17380,6 @@ 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\" or \"video\".\n",
- .syntax = "RTPQOS(<type>|<field>)",
- .read = acf_rtpqos_read,
-};
-
/*! \brief PBX load module - initialization */
static int load_module(void)
{
@@ -17443,7 +17429,6 @@ 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,
@@ -17473,7 +17458,6 @@ 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 9c02a59fa..9184cf955 100644
--- a/funcs/func_channel.c
+++ b/funcs/func_channel.c
@@ -149,18 +149,36 @@ 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"
+ " 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"