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Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to. 

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71988 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
file 2007-06-26 23:31:23 +00:00
parent d806540826
commit b401a5b5e5
3 changed files with 65 additions and 42 deletions
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1
CHANGES
View File

@ -59,6 +59,7 @@ SIP changes
required due to the restructuring of how MWI is handled. See the descriptions
in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
for more information.
* Added rtpdest option to CHANNEL() dialplan function.
IAX2 changes
------------

101
channels/chan_sip.c
View File

@ -15001,9 +15001,9 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
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);
int res = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
AST_APP_ARG(type);
@ -15017,51 +15017,68 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char
return 0;
}
if (ast_strlen_zero(args.param) || strcasecmp(args.param, "rtpqos"))
return -1;
/* Default arguments of audio,all */
if (ast_strlen_zero(args.type))
args.type = "audio";
if (ast_strlen_zero(args.field))
args.field = "all";
memset(buf, 0, buflen);
memset(&qos, 0, sizeof(qos));
if (strcasecmp(args.type, "AUDIO") == 0) {
all = ast_rtp_get_quality(p->rtp, &qos);
} else if (strcasecmp(args.type, "VIDEO") == 0) {
all = ast_rtp_get_quality(p->vrtp, &qos);
} else if (strcasecmp(args.type, "TEXT") == 0) {
all = ast_rtp_get_quality(p->trtp, &qos);
}
if (!strcasecmp(args.param, "rtpdest")) {
struct sockaddr_in sin;
if (strcasecmp(args.field, "local_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.local_ssrc);
else if (strcasecmp(args.field, "local_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.local_lostpackets);
else if (strcasecmp(args.field, "local_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
else if (strcasecmp(args.field, "local_count") == 0)
snprintf(buf, buflen, "%u", qos.local_count);
else if (strcasecmp(args.field, "remote_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.remote_ssrc);
else if (strcasecmp(args.field, "remote_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.remote_lostpackets);
else if (strcasecmp(args.field, "remote_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
else if (strcasecmp(args.field, "remote_count") == 0)
snprintf(buf, buflen, "%u", qos.remote_count);
else if (strcasecmp(args.field, "rtt") == 0)
snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
else if (strcasecmp(args.field, "all") == 0)
ast_copy_string(buf, all, buflen);
else {
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
return -1;
if (ast_strlen_zero(args.type))
args.type = "audio";
if (!strcasecmp(args.type, "audio"))
ast_rtp_get_peer(p->rtp, &sin);
else if (!strcasecmp(args.type, "video"))
ast_rtp_get_peer(p->vrtp, &sin);
else if (!strcasecmp(args.type, "text"))
ast_rtp_get_peer(p->trtp, &sin);
snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else if (!strcasecmp(args.param, "rtpqos")) {
struct ast_rtp_quality qos;
memset(&qos, 0, sizeof(qos));
if (ast_strlen_zero(args.type))
args.type = "audio";
if (ast_strlen_zero(args.field))
args.field = "all";
if (strcasecmp(args.type, "AUDIO") == 0) {
all = ast_rtp_get_quality(p->rtp, &qos);
} else if (strcasecmp(args.type, "VIDEO") == 0) {
all = ast_rtp_get_quality(p->vrtp, &qos);
} else if (strcasecmp(args.type, "TEXT") == 0) {
all = ast_rtp_get_quality(p->trtp, &qos);
}
if (strcasecmp(args.field, "local_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.local_ssrc);
else if (strcasecmp(args.field, "local_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.local_lostpackets);
else if (strcasecmp(args.field, "local_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
else if (strcasecmp(args.field, "local_count") == 0)
snprintf(buf, buflen, "%u", qos.local_count);
else if (strcasecmp(args.field, "remote_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.remote_ssrc);
else if (strcasecmp(args.field, "remote_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.remote_lostpackets);
else if (strcasecmp(args.field, "remote_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
else if (strcasecmp(args.field, "remote_count") == 0)
snprintf(buf, buflen, "%u", qos.remote_count);
else if (strcasecmp(args.field, "rtt") == 0)
snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
else if (strcasecmp(args.field, "all") == 0)
ast_copy_string(buf, all, buflen);
else {
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
return -1;
}
} else {
res = -1;
}
return 0;
return res;
}
/*! \brief Handle incoming BYE request */

5
funcs/func_channel.c
View File

@ -181,6 +181,11 @@ static struct ast_custom_function channel_function = {
" 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"
"R/O rtpdest Get remote RTP destination information\n"
" This option takes one additional argument:\n"
" Argument 1:\n"
" audio Get audio destination\n"
" video Get video destination\n"
"\n"
"chan_iax2 provides the following additional options:\n"
"R/W osptoken Get or set the OSP token information for a call\n"