kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-03 11:25:35 +00:00
Code Issues Packages Projects Releases Wiki Activity

ACN: Add tracing to existing code

Browse Source 
Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.

No functional changes were made with this commit.

Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c
This commit is contained in:
George Joseph
2020-07-06 13:23:24 -06:00
parent 2d22e34206
commit 9bd1d686a1
9 changed files with 438 additions and 179 deletions
Download Patch File Download Diff File Expand all files Collapse all files

80
res/res_pjsip_sdp_rtp.c
View File

@@ -324,6 +324,7 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
char fmt_param[256];
enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
AST_RTP_OPT_G726_NONSTANDARD : 0;
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
ast_rtp_codecs_payloads_initialize(codecs);
@@ -395,6 +396,8 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
ast_rtp_codecs_set_framing(codecs, framing);
}
}
SCOPE_EXIT_RTN();
}
static int apply_cap_to_bundled(struct ast_sip_session_media *session_media,
@@ -436,12 +439,14 @@ static struct ast_format_cap *set_incoming_call_offer_cap(
struct ast_format_cap *remote;
struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
int fmts = 0;
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
remote = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!remote) {
ast_log(LOG_ERROR, "Failed to allocate %s incoming remote capabilities\n",
ast_codec_media_type2str(session_media->type));
return NULL;
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't allocate caps\n");
}
/* Get the peer's capabilities*/
@@ -456,7 +461,7 @@ static struct ast_format_cap *set_incoming_call_offer_cap(
if (!incoming_call_offer_cap || ast_format_cap_empty(incoming_call_offer_cap)) {
ao2_cleanup(incoming_call_offer_cap);
ast_rtp_codecs_payloads_destroy(&codecs);
return NULL;
SCOPE_EXIT_RTN_VALUE(NULL, "No incoming call offer caps\n");
}
/*
@@ -470,7 +475,7 @@ static struct ast_format_cap *set_incoming_call_offer_cap(
ast_rtp_codecs_payloads_destroy(&codecs);
return incoming_call_offer_cap;
SCOPE_EXIT_RTN_VALUE(incoming_call_offer_cap);
}
static int set_caps(struct ast_sip_session *session,
@@ -488,13 +493,15 @@ static int set_caps(struct ast_sip_session *session,
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
ast_format_cap_count(session->direct_media_cap);
int dsp_features = 0;
SCOPE_ENTER(1, "%s %s\n", ast_sip_session_get_name(session), is_offer ? "OFFER" : "ANSWER");
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n",
ast_codec_media_type2str(session_media->type));
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create %s capabilities\n",
ast_codec_media_type2str(session_media->type));
}
/* get the endpoint capabilities */
@@ -519,7 +526,10 @@ static int set_caps(struct ast_sip_session *session,
ast_codec_media_type2str(session_media->type),
ast_format_cap_get_names(caps, &usbuf),
ast_format_cap_get_names(peer, &thembuf));
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
ast_codec_media_type2str(session_media->type),
ast_format_cap_get_names(caps, &usbuf),
ast_format_cap_get_names(peer, &thembuf));
}
if (is_offer) {
@@ -594,7 +604,7 @@ static int set_caps(struct ast_sip_session *session,
}
ast_rtp_codecs_payloads_destroy(&codecs);
return 0;
SCOPE_EXIT_RTN_VALUE(0);
}
static pjmedia_sdp_attr* generate_rtpmap_attr(struct ast_sip_session *session, pjmedia_sdp_media *media, pj_pool_t *pool,
@@ -1416,12 +1426,13 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
enum ast_sip_session_media_encryption encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
struct ast_format_cap *joint;
int res;
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
/* If no type formats have been configured reject this stream */
if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type)) {
ast_debug(3, "Endpoint has no codecs for media type '%s', declining stream\n",
ast_codec_media_type2str(session_media->type));
return 0;
SCOPE_EXIT_RTN_VALUE(0, "Endpoint has no codecs\n");
}
/* Ensure incoming transport is compatible with the endpoint's configuration */
@@ -1429,7 +1440,7 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
encryption = check_endpoint_media_transport(session->endpoint, stream);
if (encryption == AST_SIP_MEDIA_TRANSPORT_INVALID) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Incompatible transport\n");
}
}
@@ -1438,12 +1449,12 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
/* The provided host was actually invalid so we error out this negotiation */
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Invalid host\n");
}
/* Using the connection information create an appropriate RTP instance */
if (!session_media->rtp && create_rtp(session, session_media, sdp)) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
}
process_ssrc_attributes(session, session_media, stream);
@@ -1465,7 +1476,7 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
* but was not this session must fail. This must also fail if crypto was
* required in the offer but could not be set up.
*/
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Incompatible crypto\n");
}
/* There is no encryption, sad. */
session_media->encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
@@ -1510,10 +1521,10 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
res = apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream, joint);
ao2_cleanup(joint);
if (res != 0) {
return 0;
SCOPE_EXIT_RTN_VALUE(0, "Something failed\n");
}
return 1;
SCOPE_EXIT_RTN_VALUE(1);
}
static int add_crypto_to_stream(struct ast_sip_session *session,
@@ -1666,13 +1677,14 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
enum ast_media_type media_type = session_media->type;
struct ast_sip_session_media *session_media_transport;
pj_sockaddr ip;
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
ast_format_cap_count(session->direct_media_cap);
SCOPE_ENTER(1, "%s Type: %s %s\n", ast_sip_session_get_name(session),
ast_codec_media_type2str(media_type), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
media = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_media));
if (!media) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Pool alloc failure\n");
}
pj_strdup2(pool, &media->desc.media, ast_codec_media_type2str(session_media->type));
@@ -1702,11 +1714,11 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
sdp->media[sdp->media_count++] = media;
ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
return 1;
SCOPE_EXIT_RTN_VALUE(1, "Stream removed or no formats\n");
}
if (!session_media->rtp && create_rtp(session, session_media, sdp)) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
}
/* If this stream has not been bundled already it is new and we need to ensure there is no SSRC conflict */
@@ -1738,7 +1750,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
* set up according to the configuration. This ends up changing the media transport.
*/
if (add_crypto_to_stream(session, session_media, pool, media)) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't add crypto\n");
}
if (pj_strlen(&session_media->transport)) {
@@ -1755,7 +1767,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn));
if (!media->conn) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Pool alloc failure\n");
}
/* Add connection level details */
@@ -1770,7 +1782,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
if (ast_strlen_zero(hostip)) {
ast_log(LOG_ERROR, "No local host IP available for stream %s\n",
ast_codec_media_type2str(session_media->type));
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "No local host ip\n");
}
media->conn->net_type = STR_IN;
@@ -1798,7 +1810,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
media->desc.port = bundle_group_stream->desc.port;
if (add_crypto_to_stream(session, session_media_transport, pool, media)) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't add crypto\n");
}
add_ice_to_stream(session, session_media_transport, pool, media, 0);
@@ -1809,7 +1821,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n",
ast_codec_media_type2str(session_media->type));
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create caps\n");
}
if (direct_media_enabled) {
@@ -1896,7 +1908,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
/* If no formats were actually added to the media stream don't add it to the SDP */
if (!media->desc.fmt_count) {
return 1;
SCOPE_EXIT_RTN_VALUE(1, "No formats added to stream\n");
}
/* If ptime is set add it as an attribute */
@@ -1952,7 +1964,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
/* Add the media stream to the SDP */
sdp->media[sdp->media_count++] = media;
return 1;
SCOPE_EXIT_RTN_VALUE(1, "RC: 1\n");
}
static struct ast_frame *media_session_rtp_read_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
@@ -2010,20 +2022,22 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
char host[NI_MAXHOST];
int res;
struct ast_sip_session_media *session_media_transport;
SCOPE_ENTER(1, "%s Stream: %s\n", ast_sip_session_get_name(session),
ast_str_tmp(128, ast_stream_to_str(asterisk_stream, &STR_TMP)));
if (!session->channel) {
return 1;
SCOPE_EXIT_RTN_VALUE(1, "No channel\n");
}
/* Ensure incoming transport is compatible with the endpoint's configuration */
if (!session->endpoint->media.rtp.use_received_transport &&
check_endpoint_media_transport(session->endpoint, remote_stream) == AST_SIP_MEDIA_TRANSPORT_INVALID) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Incompatible transport\n");
}
/* Create an RTP instance if need be */
if (!session_media->rtp && create_rtp(session, session_media, local)) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
}
process_ssrc_attributes(session, session_media, remote_stream);
@@ -2042,11 +2056,11 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
/* If optimistic encryption is disabled and crypto should have been enabled but was not
* this session must fail.
*/
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Incompatible crypto\n");
}
if (!remote_stream->conn && !remote->conn) {
return 1;
SCOPE_EXIT_RTN_VALUE(1, "No connection info\n");
}
ast_copy_pj_str(host, remote_stream->conn ? &remote_stream->conn->addr : &remote->conn->addr, sizeof(host));
@@ -2054,7 +2068,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
/* The provided host was actually invalid so we error out this negotiation */
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "Host invalid\n");
}
/* Apply connection information to the RTP instance */
@@ -2079,7 +2093,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
}
if (set_caps(session, session_media, session_media_transport, remote_stream, 0, asterisk_stream)) {
return -1;
SCOPE_EXIT_RTN_VALUE(-1, "set_caps failed\n");
}
/* Set the channel uniqueid on the RTP instance now that it is becoming active */
@@ -2097,7 +2111,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
}
return 1;
SCOPE_EXIT_RTN_VALUE(1, "moh\n");
}
if (session_media->remotely_held_changed) {
@@ -2155,7 +2169,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
session_media, 1);
}
return 1;
SCOPE_EXIT_RTN_VALUE(1, "Handled\n");
}
/*! \brief Function which updates the media stream with external media address, if applicable */