[svn-commits] mjordan: trunk r419825 - in /trunk: ./ channels/ main/ res/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jul 31 06:57:53 CDT 2014
Author: mjordan
Date: Thu Jul 31 06:57:51 2014
New Revision: 419825
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=419825
Log:
res_hep_rtcp: Add module that sends RTCP information to a Homer Server
This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes
to the RTCP topics in Stasis and receives RTCP information back from the
message bus. It encodes into HEPv3 packets and sends the information to the
res_hep module for transmission.
Using this, someone with a Homer server can get live call quality monitoring
for all RTP-based channels in their Asterisk 12+ systems.
In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and
chan_pjsip that were uncovered by the tests written for the Asterisk Test
Suite. This patch fixes the following:
1) chan_pjsip failed to set its channel unique ids on its RTP instance on
outbound calls. It now does this in the appropriate location, in the
serialized call callback.
2) The rtp_engine was overflowing some values when packed into JSON.
Specifically, some longs and unsigned ints can't be be packed into integer
values, for obvious reasons. Since libjansson only supports integers,
floats, strings, booleans, and objects, we print these values into strings.
3) res_rtp_asterisk had a few problems:
(a) it would emit a source IP address of 0.0.0.0 if bound to that IP
address. We now use ast_find_ourip to get a better IP address, and
properly marshal the result into an ast_strdupa'd string.
(b) Reports can be generated with no report bodies. In particular, this
occurs when a sender is transmitting information to a receiver (who
will send no RTP back to the sender). As such, the sender has no report
body for what it received. We now properly handle this case, and the
sender will emit SR reports with no body. Likewise, if we receive an
RTCP packet with no report body, we will still generate the appropriate
events.
ASTERISK-24119 #close
........
Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12
Added:
trunk/res/res_hep_rtcp.c
- copied unchanged from r419823, branches/12/res/res_hep_rtcp.c
Modified:
trunk/ (props changed)
trunk/CHANGES
trunk/channels/chan_pjsip.c
trunk/main/rtp_engine.c
trunk/res/res_rtp_asterisk.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-12-merged' - no diff available.
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=419825&r1=419824&r2=419825
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Thu Jul 31 06:57:51 2014
@@ -244,6 +244,11 @@
* The endpoint configuration object now supports 'accountcode'. Any channel
created for an endpoint with this setting will have its accountcode set
to the specified value.
+
+res_hep_rtcp
+------------------
+ * A new module, res_hep_rtcp, has been added that will forward RTCP call
+ statistics to a HEP capture server. See res_hep for more information.
Functions
------------------
Modified: trunk/channels/chan_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_pjsip.c?view=diff&rev=419825&r1=419824&r2=419825
==============================================================================
--- trunk/channels/chan_pjsip.c (original)
+++ trunk/channels/chan_pjsip.c Thu Jul 31 06:57:51 2014
@@ -353,6 +353,16 @@
.update_peer = chan_pjsip_set_rtp_peer,
};
+static void set_channel_on_rtp_instance(struct chan_pjsip_pvt *pvt, const char *channel_id)
+{
+ if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
+ ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, channel_id);
+ }
+ if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
+ ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, channel_id);
+ }
+}
+
/*! \brief Function called to create a new PJSIP Asterisk channel */
static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
{
@@ -452,12 +462,7 @@
* these will need to be recaptured as well */
pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
- if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
- ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
- }
- if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
- ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
- }
+ set_channel_on_rtp_instance(pvt, ast_channel_uniqueid(chan));
return chan;
}
@@ -685,12 +690,7 @@
struct chan_pjsip_pvt *pvt = channel->pvt;
channel->session->channel = fix_data->chan;
- if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
- ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
- }
- if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
- ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(fix_data->chan));
- }
+ set_channel_on_rtp_instance(pvt, ast_channel_uniqueid(fix_data->chan));
return 0;
}
@@ -1523,7 +1523,9 @@
static int call(void *data)
{
- struct ast_sip_session *session = data;
+ struct ast_sip_channel_pvt *channel = data;
+ struct ast_sip_session *session = channel->session;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
pjsip_tx_data *tdata;
int res = ast_sip_session_create_invite(session, &tdata);
@@ -1532,10 +1534,11 @@
ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
ast_queue_hangup(session->channel);
} else {
+ set_channel_on_rtp_instance(pvt, ast_channel_uniqueid(session->channel));
update_initial_connected_line(session);
ast_sip_session_send_request(session, tdata);
}
- ao2_ref(session, -1);
+ ao2_ref(channel, -1);
return res;
}
@@ -1544,10 +1547,10 @@
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
- ao2_ref(channel->session, +1);
- if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
+ ao2_ref(channel, +1);
+ if (ast_sip_push_task(channel->session->serializer, call, channel)) {
ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
- ao2_cleanup(channel->session);
+ ao2_cleanup(channel);
return -1;
}
@@ -1632,12 +1635,7 @@
static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct chan_pjsip_pvt *pvt)
{
session->channel = NULL;
- if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
- ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, "");
- }
- if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
- ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, "");
- }
+ set_channel_on_rtp_instance(pvt, "");
ast_channel_tech_pvt_set(ast, NULL);
}
Modified: trunk/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/rtp_engine.c?view=diff&rev=419825&r1=419824&r2=419825
==============================================================================
--- trunk/main/rtp_engine.c (original)
+++ trunk/main/rtp_engine.c Thu Jul 31 06:57:51 2014
@@ -1909,13 +1909,15 @@
for (i = 0; i < payload->report->reception_report_count; i++) {
struct ast_json *json_report_block;
- json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: i, s: i}",
+ char str_lsr[32];
+ snprintf(str_lsr, sizeof(str_lsr), "%u", payload->report->report_block[i]->lsr);
+ json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: s, s: i}",
"source_ssrc", payload->report->report_block[i]->source_ssrc,
"fraction_lost", payload->report->report_block[i]->lost_count.fraction,
"packets_lost", payload->report->report_block[i]->lost_count.packets,
"highest_seq_no", payload->report->report_block[i]->highest_seq_no,
"ia_jitter", payload->report->report_block[i]->ia_jitter,
- "lsr", payload->report->report_block[i]->lsr,
+ "lsr", str_lsr,
"dlsr", payload->report->report_block[i]->dlsr);
if (!json_report_block) {
return NULL;
@@ -1927,9 +1929,13 @@
}
if (payload->report->type == AST_RTP_RTCP_SR) {
- json_rtcp_sender_info = ast_json_pack("{s: i, s: i, s: i, s: i, s: i}",
- "ntp_timestamp_sec", payload->report->sender_information.ntp_timestamp.tv_sec,
- "ntp_timestamp_usec", payload->report->sender_information.ntp_timestamp.tv_usec,
+ char sec[32];
+ char usec[32];
+ snprintf(sec, sizeof(sec), "%ld", payload->report->sender_information.ntp_timestamp.tv_sec);
+ snprintf(usec, sizeof(usec), "%ld", payload->report->sender_information.ntp_timestamp.tv_usec);
+ json_rtcp_sender_info = ast_json_pack("{s: s, s: s, s: i, s: i, s: i}",
+ "ntp_timestamp_sec", sec,
+ "ntp_timestamp_usec", usec,
"rtp_timestamp", payload->report->sender_information.rtp_timestamp,
"packets", payload->report->sender_information.packet_count,
"octets", payload->report->sender_information.octet_count);
Modified: trunk/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_rtp_asterisk.c?view=diff&rev=419825&r1=419824&r2=419825
==============================================================================
--- trunk/res/res_rtp_asterisk.c (original)
+++ trunk/res/res_rtp_asterisk.c Thu Jul 31 06:57:51 2014
@@ -2637,10 +2637,15 @@
int rate = rtp_get_rate(rtp->f.subclass.format);
int ice;
int header_offset = 0;
- struct ast_sockaddr remote_address = { {0,} };
- struct ast_rtp_rtcp_report_block *report_block;
+ char *str_remote_address;
+ char *str_local_address;
+ struct ast_sockaddr remote_address = { { 0, } };
+ struct ast_sockaddr local_address = { { 0, } };
+ struct ast_sockaddr real_remote_address = { { 0, } };
+ struct ast_sockaddr real_local_address = { { 0, } };
+ struct ast_rtp_rtcp_report_block *report_block = NULL;
RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report,
- ast_rtp_rtcp_report_alloc(1),
+ ast_rtp_rtcp_report_alloc(rtp->themssrc ? 1 : 0),
ao2_cleanup);
if (!rtp || !rtp->rtcp) {
@@ -2656,16 +2661,11 @@
return 1;
}
- report_block = ast_calloc(1, sizeof(*report_block));
- if (!report_block) {
- return 1;
- }
-
/* Compute statistics */
calculate_lost_packet_statistics(rtp, &lost_packets, &fraction_lost);
gettimeofday(&now, NULL);
- rtcp_report->reception_report_count = 1;
+ rtcp_report->reception_report_count = rtp->themssrc ? 1 : 0;
rtcp_report->ssrc = rtp->ssrc;
rtcp_report->type = sr ? RTCP_PT_SR : RTCP_PT_RR;
if (sr) {
@@ -2674,17 +2674,25 @@
rtcp_report->sender_information.packet_count = rtp->txcount;
rtcp_report->sender_information.octet_count = rtp->txoctetcount;
}
- rtcp_report->report_block[0] = report_block;
- report_block->source_ssrc = rtp->themssrc;
- report_block->lost_count.fraction = (fraction_lost & 0xff);
- report_block->lost_count.packets = (lost_packets & 0xffffff);
- report_block->highest_seq_no = (rtp->cycles | (rtp->lastrxseqno & 0xffff));
- report_block->ia_jitter = (unsigned int)(rtp->rxjitter * rate);
- report_block->lsr = rtp->rtcp->themrxlsr;
- /* If we haven't received an SR report, DLSR should be 0 */
- if (!ast_tvzero(rtp->rtcp->rxlsr)) {
- timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
- report_block->dlsr = (((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000;
+
+ if (rtp->themssrc) {
+ report_block = ast_calloc(1, sizeof(*report_block));
+ if (!report_block) {
+ return 1;
+ }
+
+ rtcp_report->report_block[0] = report_block;
+ report_block->source_ssrc = rtp->themssrc;
+ report_block->lost_count.fraction = (fraction_lost & 0xff);
+ report_block->lost_count.packets = (lost_packets & 0xffffff);
+ report_block->highest_seq_no = (rtp->cycles | (rtp->lastrxseqno & 0xffff));
+ report_block->ia_jitter = (unsigned int)(rtp->rxjitter * rate);
+ report_block->lsr = rtp->rtcp->themrxlsr;
+ /* If we haven't received an SR report, DLSR should be 0 */
+ if (!ast_tvzero(rtp->rtcp->rxlsr)) {
+ timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
+ report_block->dlsr = (((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000;
+ }
}
timeval2ntp(rtcp_report->sender_information.ntp_timestamp, &now_msw, &now_lsw);
rtcpheader = (unsigned int *)bdata;
@@ -2699,14 +2707,17 @@
rtcpheader[6] = htonl(rtcp_report->sender_information.octet_count);
len += 20;
}
- rtcpheader[2 + header_offset] = htonl(report_block->source_ssrc); /* Their SSRC */
- rtcpheader[3 + header_offset] = htonl((report_block->lost_count.fraction << 24) | report_block->lost_count.packets);
- rtcpheader[4 + header_offset] = htonl(report_block->highest_seq_no);
- rtcpheader[5 + header_offset] = htonl(report_block->ia_jitter);
- rtcpheader[6 + header_offset] = htonl(report_block->lsr);
- rtcpheader[7 + header_offset] = htonl(report_block->dlsr);
- len += 24;
- rtcpheader[0] = htonl((2 << 30) | (1 << 24) | ((sr ? RTCP_PT_SR : RTCP_PT_RR) << 16) | ((len/4)-1));
+ if (report_block) {
+ rtcpheader[2 + header_offset] = htonl(report_block->source_ssrc); /* Their SSRC */
+ rtcpheader[3 + header_offset] = htonl((report_block->lost_count.fraction << 24) | report_block->lost_count.packets);
+ rtcpheader[4 + header_offset] = htonl(report_block->highest_seq_no);
+ rtcpheader[5 + header_offset] = htonl(report_block->ia_jitter);
+ rtcpheader[6 + header_offset] = htonl(report_block->lsr);
+ rtcpheader[7 + header_offset] = htonl(report_block->dlsr);
+ len += 24;
+ }
+ rtcpheader[0] = htonl((2 << 30) | (rtcp_report->reception_report_count << 24)
+ | ((sr ? RTCP_PT_SR : RTCP_PT_RR) << 16) | ((len/4)-1));
/* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
/* it can change mid call, and SDES can't) */
@@ -2758,8 +2769,22 @@
ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(report_block->dlsr / 65536.0));
}
- message_blob = ast_json_pack("{s: s}",
- "to", ast_sockaddr_stringify(&remote_address));
+ ast_rtp_instance_get_local_address(instance, &local_address);
+ if (!ast_find_ourip(&real_local_address, &local_address, 0)) {
+ str_local_address = ast_strdupa(ast_sockaddr_stringify(&real_local_address));
+ } else {
+ str_local_address = ast_strdupa(ast_sockaddr_stringify(&local_address));
+ }
+
+ if (!ast_find_ourip(&real_remote_address, &remote_address, 0)) {
+ str_remote_address = ast_strdupa(ast_sockaddr_stringify(&real_remote_address));
+ } else {
+ str_remote_address = ast_strdupa(ast_sockaddr_stringify(&remote_address));
+ }
+
+ message_blob = ast_json_pack("{s: s, s: s}",
+ "to", str_remote_address,
+ "from", str_local_address);
ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_sent_type(),
rtcp_report,
message_blob);
@@ -3574,6 +3599,11 @@
int report_counter = 0;
struct ast_rtp_rtcp_report_block *report_block;
struct ast_frame *f = &ast_null_frame;
+ char *str_local_address;
+ char *str_remote_address;
+ struct ast_sockaddr local_address = { { 0,} };
+ struct ast_sockaddr real_local_address = { { 0, } };
+ struct ast_sockaddr real_remote_address = { { 0, } };
/* Read in RTCP data from the socket */
if ((res = rtcp_recvfrom(instance, rtcpdata + AST_FRIENDLY_OFFSET,
@@ -3630,6 +3660,8 @@
ast_debug(1, "Got RTCP report of %d bytes\n", res);
+ ast_rtp_instance_get_local_address(instance, &local_address);
+
while (position < packetwords) {
int i, pt, rc;
unsigned int length;
@@ -3667,11 +3699,6 @@
}
i += 2; /* Advance past header and ssrc */
- if (rc == 0 && pt == RTCP_PT_RR) {
- /* We're receiving a receiver report with no reports, which is ok */
- position += (length + 1);
- continue;
- }
switch (pt) {
case RTCP_PT_SR:
gettimeofday(&rtp->rtcp->rxlsr, NULL);
@@ -3696,64 +3723,75 @@
rtcp_report->sender_information.octet_count);
}
i += 5;
- if (rc < 1) {
- break;
- }
/* Intentional fall through */
case RTCP_PT_RR:
if (rtcp_report->type != RTCP_PT_SR) {
rtcp_report->type = RTCP_PT_RR;
}
- /* Don't handle multiple reception reports (rc > 1) yet */
- report_block = ast_calloc(1, sizeof(*report_block));
- if (!report_block) {
- return &ast_null_frame;
+ if (rc > 0) {
+ /* Don't handle multiple reception reports (rc > 1) yet */
+ report_block = ast_calloc(1, sizeof(*report_block));
+ if (!report_block) {
+ return &ast_null_frame;
+ }
+ rtcp_report->report_block[report_counter] = report_block;
+ report_block->source_ssrc = ntohl(rtcpheader[i]);
+ report_block->lost_count.packets = ntohl(rtcpheader[i + 1]) & 0x00ffffff;
+ report_block->lost_count.fraction = ((ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24);
+ report_block->highest_seq_no = ntohl(rtcpheader[i + 2]);
+ report_block->ia_jitter = ntohl(rtcpheader[i + 3]);
+ report_block->lsr = ntohl(rtcpheader[i + 4]);
+ report_block->dlsr = ntohl(rtcpheader[i + 5]);
+ if (report_block->lsr
+ && update_rtt_stats(rtp, report_block->lsr, report_block->dlsr)
+ && rtcp_debug_test_addr(&addr)) {
+ struct timeval now;
+ unsigned int lsr_now, lsw, msw;
+ gettimeofday(&now, NULL);
+ timeval2ntp(now, &msw, &lsw);
+ lsr_now = (((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16));
+ ast_verbose("Internal RTCP NTP clock skew detected: "
+ "lsr=%u, now=%u, dlsr=%u (%u:%03ums), "
+ "diff=%u\n",
+ report_block->lsr, lsr_now, report_block->dlsr, report_block->dlsr / 65536,
+ (report_block->dlsr % 65536) * 1000 / 65536,
+ report_block->dlsr - (lsr_now - report_block->lsr));
+ }
+ update_jitter_stats(rtp, report_block->ia_jitter);
+ update_lost_stats(rtp, report_block->lost_count.packets);
+ rtp->rtcp->reported_jitter_count++;
+
+ if (rtcp_debug_test_addr(&addr)) {
+ ast_verbose(" Fraction lost: %d\n", report_block->lost_count.fraction);
+ ast_verbose(" Packets lost so far: %u\n", report_block->lost_count.packets);
+ ast_verbose(" Highest sequence number: %u\n", report_block->highest_seq_no & 0x0000ffff);
+ ast_verbose(" Sequence number cycles: %u\n", report_block->highest_seq_no >> 16);
+ ast_verbose(" Interarrival jitter: %u\n", report_block->ia_jitter);
+ ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long)(report_block->lsr) >> 16,((unsigned long)(report_block->lsr) << 16) * 4096);
+ ast_verbose(" DLSR: %4.4f (sec)\n",(double)report_block->dlsr / 65536.0);
+ ast_verbose(" RTT: %4.4f(sec)\n", rtp->rtcp->rtt);
+ }
+ report_counter++;
}
- rtcp_report->report_block[report_counter] = report_block;
- report_block->source_ssrc = ntohl(rtcpheader[i]);
- report_block->lost_count.packets = ntohl(rtcpheader[i + 1]) & 0x00ffffff;
- report_block->lost_count.fraction = ((ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24);
- report_block->highest_seq_no = ntohl(rtcpheader[i + 2]);
- report_block->ia_jitter = ntohl(rtcpheader[i + 3]);
- report_block->lsr = ntohl(rtcpheader[i + 4]);
- report_block->dlsr = ntohl(rtcpheader[i + 5]);
- if (report_block->lsr
- && update_rtt_stats(rtp, report_block->lsr, report_block->dlsr)
- && rtcp_debug_test_addr(&addr)) {
- struct timeval now;
- unsigned int lsr_now, lsw, msw;
- gettimeofday(&now, NULL);
- timeval2ntp(now, &msw, &lsw);
- lsr_now = (((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16));
- ast_verbose("Internal RTCP NTP clock skew detected: "
- "lsr=%u, now=%u, dlsr=%u (%u:%03ums), "
- "diff=%u\n",
- report_block->lsr, lsr_now, report_block->dlsr, report_block->dlsr / 65536,
- (report_block->dlsr % 65536) * 1000 / 65536,
- report_block->dlsr - (lsr_now - report_block->lsr));
- }
- update_jitter_stats(rtp, report_block->ia_jitter);
- update_lost_stats(rtp, report_block->lost_count.packets);
- rtp->rtcp->reported_jitter_count++;
-
- if (rtcp_debug_test_addr(&addr)) {
- ast_verbose(" Fraction lost: %d\n", report_block->lost_count.fraction);
- ast_verbose(" Packets lost so far: %u\n", report_block->lost_count.packets);
- ast_verbose(" Highest sequence number: %u\n", report_block->highest_seq_no & 0x0000ffff);
- ast_verbose(" Sequence number cycles: %u\n", report_block->highest_seq_no >> 16);
- ast_verbose(" Interarrival jitter: %u\n", report_block->ia_jitter);
- ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long)(report_block->lsr) >> 16,((unsigned long)(report_block->lsr) << 16) * 4096);
- ast_verbose(" DLSR: %4.4f (sec)\n",(double)report_block->dlsr / 65536.0);
- ast_verbose(" RTT: %4.4f(sec)\n", rtp->rtcp->rtt);
- }
- report_counter++;
-
/* If and when we handle more than one report block, this should occur outside
* this loop.
*/
- message_blob = ast_json_pack("{s: s, s: f}",
- "from", ast_sockaddr_stringify(&addr),
+ if (!ast_find_ourip(&real_local_address, &local_address, 0)) {
+ str_local_address = ast_strdupa(ast_sockaddr_stringify(&real_local_address));
+ } else {
+ str_local_address = ast_strdupa(ast_sockaddr_stringify(&local_address));
+ }
+
+ if (!ast_find_ourip(&real_remote_address, &addr, 0)) {
+ str_remote_address = ast_strdupa(ast_sockaddr_stringify(&real_remote_address));
+ } else {
+ str_remote_address = ast_strdupa(ast_sockaddr_stringify(&addr));
+ }
+
+ message_blob = ast_json_pack("{s: s, s: s, s: f}",
+ "from", str_remote_address,
+ "to", str_local_address,
"rtt", rtp->rtcp->rtt);
ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_received_type(),
rtcp_report,
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