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Logging: Add debug logging categories

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Added debug logging categories that allow a user to output debug
information based on a specified category. This lets the user limit,
and filter debug output to data relevant to a particular context,
or topic. For instance the following categories are now available for
debug logging purposes:

  dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
  stun, stun_packet

These debug categories can be enable/disable via an Asterisk CLI command.

While this overrides, and outputs debug data, core system debugging is
not affected by this patch. Statements still output at their appropriate
debug level. As well backwards compatibility has been maintained with
past debug groups that could be enabled using the CLI (e.g. rtpdebug,
stundebug, etc.).

ASTERISK-29054 #close

Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849
(cherry picked from commit 56028426de)
This commit is contained in:
Kevin Harwell
2020-08-28 16:32:37 -05:00
parent a6faa53af0
commit 6255e7976c
11 changed files with 1035 additions and 200 deletions
Download Patch File Download Diff File Expand all files Collapse all files

23
res/res_pjsip_sdp_rtp.c
View File

@@ -52,6 +52,7 @@
#include "asterisk/dsp.h"
#include "asterisk/linkedlists.h" /* for AST_LIST_NEXT */
#include "asterisk/stream.h"
#include "asterisk/logger_category.h"
#include "asterisk/format_cache.h"
#include "asterisk/res_pjsip.h"
@@ -82,15 +83,15 @@ static int send_keepalive(const void *data)
keepalive = ast_rtp_instance_get_keepalive(rtp);
if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
ast_debug(3, "Not sending RTP keepalive on RTP instance %p since direct media is in use\n", rtp);
ast_debug_rtp(3, "(%p) RTP not sending keepalive since direct media is in use\n", rtp);
return keepalive * 1000;
}
interval = time(NULL) - ast_rtp_instance_get_last_tx(rtp);
send_keepalive = interval >= keepalive;
ast_debug(3, "It has been %d seconds since RTP was last sent on instance %p. %sending keepalive\n",
(int) interval, rtp, send_keepalive ? "S" : "Not s");
ast_debug_rtp(3, "(%p) RTP it has been %d seconds since RTP was last sent. %sending keepalive\n",
rtp, (int) interval, send_keepalive ? "S" : "Not s");
if (send_keepalive) {
ast_rtp_instance_sendcng(rtp, 0);
@@ -138,8 +139,8 @@ static int rtp_check_timeout(const void *data)
* - disconnect channel unless direct media is in use.
*/
if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
ast_debug(3, "Not disconnecting channel '%s' for lack of %s RTP activity in %d seconds "
"since direct media is in use\n", ast_channel_name(chan),
ast_debug_rtp(3, "(%p) RTP not disconnecting channel '%s' for lack of %s RTP activity in %d seconds "
"since direct media is in use\n", rtp, ast_channel_name(chan),
ast_codec_media_type2str(session_media->type), elapsed);
ast_channel_unlock(chan);
ast_channel_unref(chan);
@@ -233,12 +234,12 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
if (ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0)) {
ast_debug(1, "Endpoint %s: Binding RTP media to %s\n",
ast_debug_rtp(1, "Endpoint %s: Binding RTP media to %s\n",
ast_sorcery_object_get_id(session->endpoint),
session->endpoint->media.address);
media_address = &temp_media_address;
} else {
ast_debug(1, "Endpoint %s: RTP media address invalid: %s\n",
ast_debug_rtp(1, "Endpoint %s: RTP media address invalid: %s\n",
ast_sorcery_object_get_id(session->endpoint),
session->endpoint->media.address);
}
@@ -256,11 +257,11 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
pj_sockaddr_print(&trans_state->host, hoststr, sizeof(hoststr), 0);
if (ast_sockaddr_parse(&temp_media_address, hoststr, 0)) {
ast_debug(1, "Transport %s bound to %s: Using it for RTP media.\n",
ast_debug_rtp(1, "Transport %s bound to %s: Using it for RTP media.\n",
session->endpoint->transport, hoststr);
media_address = &temp_media_address;
} else {
ast_debug(1, "Transport %s bound to %s: Invalid for RTP media.\n",
ast_debug_rtp(1, "Transport %s bound to %s: Invalid for RTP media.\n",
session->endpoint->transport, hoststr);
}
ao2_ref(trans_state, -1);
@@ -786,6 +787,8 @@ static void process_ice_attributes(struct ast_sip_session *session, struct ast_s
return;
}
ast_debug_ice(2, "(%p) ICE process attributes\n", session_media->rtp);
attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-ufrag", NULL);
if (!attr) {
attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-ufrag", NULL);
@@ -794,6 +797,7 @@ static void process_ice_attributes(struct ast_sip_session *session, struct ast_s
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
ice->set_authentication(session_media->rtp, attr_value, NULL);
} else {
ast_debug_ice(2, "(%p) ICE no, or invalid ice-ufrag\n", session_media->rtp);
return;
}
@@ -805,6 +809,7 @@ static void process_ice_attributes(struct ast_sip_session *session, struct ast_s
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
ice->set_authentication(session_media->rtp, NULL, attr_value);
} else {
ast_debug_ice(2, "(%p) ICE no, or invalid ice-pwd\n", session_media->rtp);
return;
}