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Commit 0e0600a4 authored by Tilghman Lesher's avatar Tilghman Lesher
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Merged revisions 59227 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines

Change this to a single dp function to make oej happy.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
parent d6943624
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with 31 additions and 66 deletions
channels/chan_sip.c
+ 31
− 66
View file @ 0e0600a4
......@@ -14800,11 +14800,16 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
}
}
 
static int acf_audiortpqos_read(struct ast_channel *chan, const char *funcname, char *args, char *buf, size_t buflen)
static int acf_rtpqos_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 = "";
char *all = "", *parse = ast_strdupa(preparse);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(type);
AST_APP_ARG(field);
);
AST_STANDARD_APP_ARGS(args, parse);
 
/* Sanity check */
if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) {
......@@ -14814,36 +14819,36 @@ static int acf_audiortpqos_read(struct ast_channel *chan, const char *funcname,
memset(buf, 0, buflen);
memset(&qos, 0, sizeof(qos));
 
if (strcmp(funcname, "RTPAUDIOQOS") == 0) {
if (strcasecmp(args.type, "AUDIO") == 0) {
all = ast_rtp_get_quality(p->rtp, &qos);
} else if (strcmp(funcname, "RTPVIDEOQOS") == 0) {
} else if (strcasecmp(args.type, "VIDEO") == 0) {
all = ast_rtp_get_quality(p->vrtp, &qos);
} else if (strcmp(funcname, "RTPTEXTQOS") == 0) {
} else if (strcasecmp(args.type, "TEXT") == 0) {
all = ast_rtp_get_quality(p->trtp, &qos);
}
 
if (strcasecmp(args, "local_ssrc") == 0)
if (strcasecmp(args.field, "local_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.local_ssrc);
else if (strcasecmp(args, "local_lostpackets") == 0)
else if (strcasecmp(args.field, "local_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.local_lostpackets);
else if (strcasecmp(args, "local_jitter") == 0)
else if (strcasecmp(args.field, "local_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
else if (strcasecmp(args, "local_count") == 0)
else if (strcasecmp(args.field, "local_count") == 0)
snprintf(buf, buflen, "%u", qos.local_count);
else if (strcasecmp(args, "remote_ssrc") == 0)
else if (strcasecmp(args.field, "remote_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.remote_ssrc);
else if (strcasecmp(args, "remote_lostpackets") == 0)
else if (strcasecmp(args.field, "remote_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.remote_lostpackets);
else if (strcasecmp(args, "remote_jitter") == 0)
else if (strcasecmp(args.field, "remote_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
else if (strcasecmp(args, "remote_count") == 0)
else if (strcasecmp(args.field, "remote_count") == 0)
snprintf(buf, buflen, "%u", qos.remote_count);
else if (strcasecmp(args, "rtt") == 0)
else if (strcasecmp(args.field, "rtt") == 0)
snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
else if (strcasecmp(args, "all") == 0)
else if (strcasecmp(args.field, "all") == 0)
ast_copy_string(buf, all, buflen);
else {
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", args, funcname);
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
return -1;
}
return 0;
......@@ -18130,47 +18135,9 @@ static struct ast_cli_entry cli_sip[] = {
sip_reload_usage },
};
 
struct ast_custom_function acf_audiortpqos = {
.name = "RTPAUDIOQOS",
.synopsis = "Retrieve statistics about an RTP audio 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)",
.syntax = "RTPAUDIOQOS(<field>)",
.read = acf_audiortpqos_read,
};
struct ast_custom_function acf_videortpqos = {
.name = "RTPVIDEOQOS",
.synopsis = "Retrieve statistics about an RTP video 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)",
.syntax = "RTPVIDEOQOS(<field>)",
.read = acf_audiortpqos_read,
};
struct ast_custom_function acf_textrtpqos = {
.name = "RTPTEXTQOS",
.synopsis = "Retrieve statistics about an RTP text stream",
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"
......@@ -18182,9 +18149,11 @@ struct ast_custom_function acf_textrtpqos = {
" 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)",
.syntax = "RTPTEXTQOS(<field>)",
.read = acf_audiortpqos_read,
" 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 */
......@@ -18236,9 +18205,7 @@ 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_audiortpqos);
ast_custom_function_register(&acf_videortpqos);
ast_custom_function_register(&acf_textrtpqos);
ast_custom_function_register(&acf_rtpqos);
 
/* Register manager commands */
ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers,
......@@ -18268,9 +18235,7 @@ 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_audiortpqos);
ast_custom_function_unregister(&acf_videortpqos);
ast_custom_function_unregister(&acf_textrtpqos);
ast_custom_function_unregister(&acf_rtpqos);
 
/* Unregister dial plan applications */
ast_unregister_application(app_dtmfmode);
......
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