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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Tilghman Lesher
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@ -14800,11 +14800,16 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
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}
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}
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static int acf_audiortpqos_read(struct ast_channel *chan, const char *funcname, char *args, char *buf, size_t buflen)
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static int acf_rtpqos_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
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{
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struct ast_rtp_quality qos;
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struct sip_pvt *p = chan->tech_pvt;
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char *all = "";
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char *all = "", *parse = ast_strdupa(preparse);
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(type);
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AST_APP_ARG(field);
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);
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AST_STANDARD_APP_ARGS(args, parse);
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/* Sanity check */
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if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) {
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@ -14814,36 +14819,36 @@ static int acf_audiortpqos_read(struct ast_channel *chan, const char *funcname,
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memset(buf, 0, buflen);
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memset(&qos, 0, sizeof(qos));
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if (strcmp(funcname, "RTPAUDIOQOS") == 0) {
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if (strcasecmp(args.type, "AUDIO") == 0) {
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all = ast_rtp_get_quality(p->rtp, &qos);
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} else if (strcmp(funcname, "RTPVIDEOQOS") == 0) {
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} else if (strcasecmp(args.type, "VIDEO") == 0) {
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all = ast_rtp_get_quality(p->vrtp, &qos);
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} else if (strcmp(funcname, "RTPTEXTQOS") == 0) {
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} else if (strcasecmp(args.type, "TEXT") == 0) {
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all = ast_rtp_get_quality(p->trtp, &qos);
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}
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if (strcasecmp(args, "local_ssrc") == 0)
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if (strcasecmp(args.field, "local_ssrc") == 0)
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snprintf(buf, buflen, "%u", qos.local_ssrc);
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else if (strcasecmp(args, "local_lostpackets") == 0)
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else if (strcasecmp(args.field, "local_lostpackets") == 0)
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snprintf(buf, buflen, "%u", qos.local_lostpackets);
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else if (strcasecmp(args, "local_jitter") == 0)
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else if (strcasecmp(args.field, "local_jitter") == 0)
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snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
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else if (strcasecmp(args, "local_count") == 0)
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else if (strcasecmp(args.field, "local_count") == 0)
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snprintf(buf, buflen, "%u", qos.local_count);
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else if (strcasecmp(args, "remote_ssrc") == 0)
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else if (strcasecmp(args.field, "remote_ssrc") == 0)
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snprintf(buf, buflen, "%u", qos.remote_ssrc);
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else if (strcasecmp(args, "remote_lostpackets") == 0)
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else if (strcasecmp(args.field, "remote_lostpackets") == 0)
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snprintf(buf, buflen, "%u", qos.remote_lostpackets);
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else if (strcasecmp(args, "remote_jitter") == 0)
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else if (strcasecmp(args.field, "remote_jitter") == 0)
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snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
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else if (strcasecmp(args, "remote_count") == 0)
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else if (strcasecmp(args.field, "remote_count") == 0)
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snprintf(buf, buflen, "%u", qos.remote_count);
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else if (strcasecmp(args, "rtt") == 0)
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else if (strcasecmp(args.field, "rtt") == 0)
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snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
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else if (strcasecmp(args, "all") == 0)
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else if (strcasecmp(args.field, "all") == 0)
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ast_copy_string(buf, all, buflen);
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else {
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ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", args, funcname);
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ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
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return -1;
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}
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return 0;
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@ -18130,9 +18135,9 @@ static struct ast_cli_entry cli_sip[] = {
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sip_reload_usage },
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};
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struct ast_custom_function acf_audiortpqos = {
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.name = "RTPAUDIOQOS",
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.synopsis = "Retrieve statistics about an RTP audio stream",
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struct ast_custom_function acf_rtpqos = {
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.name = "RTPQOS",
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.synopsis = "Retrieve statistics about an RTP stream",
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.desc =
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"The following statistics may be retrieved:\n"
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" local_ssrc - Local SSRC (stream ID)\n"
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@ -18144,47 +18149,11 @@ struct ast_custom_function acf_audiortpqos = {
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" remote_jitter - Remote reported jitter\n"
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" remote_count - Number of transmitted packets\n"
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" rtt - Round trip time\n"
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" all - All statistics (in a form suited to logging, but not for parsing)",
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.syntax = "RTPAUDIOQOS(<field>)",
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.read = acf_audiortpqos_read,
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};
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struct ast_custom_function acf_videortpqos = {
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.name = "RTPVIDEOQOS",
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.synopsis = "Retrieve statistics about an RTP video stream",
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.desc =
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"The following statistics may be retrieved:\n"
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" local_ssrc - Local SSRC (stream ID)\n"
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" local_lostpackets - Local lost packets\n"
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" local_jitter - Local calculated jitter\n"
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" local_count - Number of received packets\n"
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" remote_ssrc - Remote SSRC (stream ID)\n"
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" remote_lostpackets - Remote lost packets\n"
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" remote_jitter - Remote reported jitter\n"
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" remote_count - Number of transmitted packets\n"
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" rtt - Round trip time\n"
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" all - All statistics (in a form suited to logging, but not for parsing)",
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.syntax = "RTPVIDEOQOS(<field>)",
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.read = acf_audiortpqos_read,
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};
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struct ast_custom_function acf_textrtpqos = {
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.name = "RTPTEXTQOS",
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.synopsis = "Retrieve statistics about an RTP text stream",
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.desc =
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"The following statistics may be retrieved:\n"
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" local_ssrc - Local SSRC (stream ID)\n"
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" local_lostpackets - Local lost packets\n"
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" local_jitter - Local calculated jitter\n"
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" local_count - Number of received packets\n"
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" remote_ssrc - Remote SSRC (stream ID)\n"
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" remote_lostpackets - Remote lost packets\n"
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" remote_jitter - Remote reported jitter\n"
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" remote_count - Number of transmitted packets\n"
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" rtt - Round trip time\n"
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" all - All statistics (in a form suited to logging, but not for parsing)",
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.syntax = "RTPTEXTQOS(<field>)",
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.read = acf_audiortpqos_read,
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" all - All statistics (in a form suited to logging, but not for parsing)\n"
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"\n"
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"Type may be specified as \"audio\", \"video\", or \"text\".\n",
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.syntax = "RTPQOS(<type>|<field>)",
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.read = acf_rtpqos_read,
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};
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/*! \brief PBX load module - initialization */
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@ -18236,9 +18205,7 @@ static int load_module(void)
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ast_custom_function_register(&sippeer_function);
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ast_custom_function_register(&sipchaninfo_function);
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ast_custom_function_register(&checksipdomain_function);
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ast_custom_function_register(&acf_audiortpqos);
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ast_custom_function_register(&acf_videortpqos);
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ast_custom_function_register(&acf_textrtpqos);
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ast_custom_function_register(&acf_rtpqos);
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/* Register manager commands */
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ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers,
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@ -18268,9 +18235,7 @@ static int unload_module(void)
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ast_custom_function_unregister(&sippeer_function);
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ast_custom_function_unregister(&sip_header_function);
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ast_custom_function_unregister(&checksipdomain_function);
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ast_custom_function_unregister(&acf_audiortpqos);
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ast_custom_function_unregister(&acf_videortpqos);
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ast_custom_function_unregister(&acf_textrtpqos);
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ast_custom_function_unregister(&acf_rtpqos);
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/* Unregister dial plan applications */
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ast_unregister_application(app_dtmfmode);
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