From 025ea9914b4f367fa9c463054475e408f1f1c0ab Mon Sep 17 00:00:00 2001
From: Yalu Zhang <yalu.zhang@iopsys.eu>
Date: Thu, 27 Jun 2024 10:21:22 +0000
Subject: [PATCH] Fix a crash when an internal call between two DECT handsets
is terminated
Use shared ubus context from the pool instead of creating a new one in endpt_get_rtp_stats().
- Do the same for endpt_signal()
- Improve the function audio_packet_handler()
- Update some log levels
---
src/channels/chan_voicemngr.c | 150 ++++++++++++++--------------------
1 file changed, 63 insertions(+), 87 deletions(-)
diff --git a/src/channels/chan_voicemngr.c b/src/channels/chan_voicemngr.c
index 188067f..9f20edd 100644
--- a/src/channels/chan_voicemngr.c
+++ b/src/channels/chan_voicemngr.c
@@ -687,29 +687,20 @@ static void endpt_signal(int line, char *signal, char *state, char *data) {
static int chan_voicemngr_send_ubus_event(char *ev_name, int line)
{
struct blob_buf blob;
- struct ubus_context *ubusctx;
int res = 0;
- ubusctx = ubus_connect(NULL);
- if (!ubusctx) {
- return -1;
- }
-
memset(&blob, 0, sizeof(blob));
if(blob_buf_init(&blob, 0)) {
- ast_ubus_free_context(ubusctx);
return -1;
}
blobmsg_add_u32(&blob, "id", line);
blobmsg_add_string(&blob, "event", ev_name);
-
- if (ubus_send_event(ubusctx, broadcast_path, blob.head) != UBUS_STATUS_OK) {
+ if (ubus_send_event(get_shared_context(__FUNCTION__), broadcast_path, blob.head) != UBUS_STATUS_OK) {
ast_log(LOG_NOTICE,"Error sending ubus message %s\n", ev_name);
res = -1;
}
- ast_ubus_free_context(ubusctx);
blob_buf_free(&blob);
return res;
@@ -1754,7 +1745,7 @@ static int chan_voicemngr_classify_rtp_packet(int payload_type) {
if (voicemngr_codecs[i].rtp_payload_type == payload_type)
return CHAN_VOICEMNGR_AUDIO;
- ast_log(LOG_WARNING, "Unknown RTP payload_type %d\n", payload_type);
+ ast_debug(3, "Unknown RTP payload_type %d\n", payload_type);
return CHAN_VOICEMNGR_UNKNOWN;
}
@@ -1764,7 +1755,7 @@ static int map_ast_codec_id_to_rtp(const struct ast_format *astcodec)
if (ast_format_cmp(astcodec, *voicemngr_codecs[i].ast_format) == AST_FORMAT_CMP_EQUAL)
return voicemngr_codecs[i].rtp_payload_type;
- ast_log(LOG_WARNING, "Unknown asterisk format(%s), return PCMA\n", ast_format_get_name(astcodec));
+ ast_debug(3, "Unknown asterisk format(%s), return PCMA\n", ast_format_get_name(astcodec));
return RTP_PT_PCMA;
}
@@ -3257,16 +3248,22 @@ static struct ast_format *map_rtpname_to_format(char* name) {
return ast_format_ulaw;
}
-/* Handle audio packets from voicemngr. */
-static void audio_packet_handler(pe_packet_t *p) {
+/* Handle audio packets from voicemngr */
+static void audio_packet_handler(pe_packet_t *p)
+{
struct chan_voicemngr_subchannel *sub;
- int packet_type = CHAN_VOICEMNGR_UNKNOWN, drop_frame = 0;
+ int packet_type = CHAN_VOICEMNGR_UNKNOWN, drop_frame = 0;
audio_packet_t *ap = (audio_packet_t *)p->data;
uint8_t payload_type = ap->rtp[1];
struct ast_frame frame = { .src = "TELCHAN", };
struct chan_voicemngr_pvt *pvt;
int sip_client_id = -1;
- unsigned int* packet_buf32 = (unsigned int*)ap->rtp;
+ unsigned int *packet_buf32 = (unsigned int*)ap->rtp;
+
+ if (!(ap->rtp_size && (ap->rtp[0] & 0x80))) {
+ ast_debug(3, "Invalid packet size or RTP version\n");
+ return;
+ }
// Clear the RTP marker bit
if ((payload_type & RTP_MARKER_BIT) && (payload_type < RTCP_SR || payload_type > RTCP_XR))
@@ -3279,77 +3276,66 @@ static void audio_packet_handler(pe_packet_t *p) {
endpt_connection(ap->line, ap->connection_id, "destroy"); // Request line close
return;
}
+
if (payload_type == sub->dtmf_pt) {
packet_type = CHAN_VOICEMNGR_AUDIO;
} else {
packet_type = chan_voicemngr_classify_rtp_packet(payload_type);
}
- //pvt_lock(sub->parent, "chan_voicemngr monitor packets");
- //ast_mutex_lock(&sub->parent->lock);
- struct ast_channel *owner = NULL;
- if (sub->owner) {
- ast_channel_ref(sub->owner);
- owner = sub->owner;
- }
- // We seem to get packets from DSP even if connection is muted (perhaps muting only affects packet callback).
- // Drop packets if subchannel is on hold. Handle rtp packet according to classification.
- if (sub->channel_state != ONHOLD && packet_type == CHAN_VOICEMNGR_AUDIO && (ap->rtp[0] & 0x80) && ap->rtp_size) {
- frame.frametype = AST_FRAME_VOICE;
- frame.offset = 0;
- frame.data.ptr = ap->rtp + 12;
- frame.datalen = ap->rtp_size - 12;
- if (payload_type == RTP_PT_CN) {
- frame.frametype = AST_FRAME_CNG;
- frame.subclass.integer = ap->rtp[12];
- } else if (sub->channel_state == OFFHOOK
- || sub->channel_state == DIALING){
- drop_frame=1;
- } else if(payload_type == sub->dtmf_pt) {
- frame.frametype = AST_FRAME_DTMF_BYPASS;
- frame.subclass.integer = ap->rtp[1]; //payload_type
- ast_debug(3, "DTMF rtp_event for digit: %d, with payload_type: %d \n", ap->rtp[12], payload_type);
- if (sub->conference_initiator == 1) {
- drop_frame=1;
- }
- } else {
- struct ast_format *format = map_rtptype_to_format(payload_type);
- if (format) {
- frame.subclass.format = format;
- frame.samples = ast_codec_samples_count(&frame);
+ enum chan_voicemngr_channel_state sub_state = sub->channel_state;
+ enum ast_channel_state owner_state = sub->owner ? ast_channel_state(sub->owner) : AST_STATE_DOWN;
+ if ((sub_state == INCALL || sub_state == CALLWAITING || sub_state == TRANSFERING) &&
+ (owner_state == AST_STATE_UP || owner_state == AST_STATE_RING)) {
+ if (packet_type == CHAN_VOICEMNGR_AUDIO) {
+ frame.frametype = AST_FRAME_VOICE;
+ frame.offset = 0;
+ frame.data.ptr = ap->rtp + 12;
+ frame.datalen = ap->rtp_size - 12;
+ if (payload_type == RTP_PT_CN) {
+ frame.frametype = AST_FRAME_CNG;
+ frame.subclass.integer = ap->rtp[12];
+ } else if(payload_type == sub->dtmf_pt) {
+ frame.frametype = AST_FRAME_DTMF_BYPASS;
+ frame.subclass.integer = ap->rtp[1]; //payload_type
+ ast_debug(3, "DTMF rtp_event for digit: %d, with payload_type: %d \n", ap->rtp[12], payload_type);
+ if (sub->conference_initiator == 1) {
+ drop_frame = 1;
+ }
+ } else {
+ struct ast_format *format = map_rtptype_to_format(payload_type);
+ if (format) {
+ frame.subclass.format = format;
+ frame.samples = ast_codec_samples_count(&frame);
+ }
}
- }
- sip_client_id = chan_voicemngr_get_sip_client_id(sub);
- if (sip_client_id >= 0 && sip_client_id < MAX_SIP_CLIENTS) {
- line_stats[sip_client_id].txpkts++;
- line_stats[sip_client_id].txbytes += frame.datalen;
+ sip_client_id = chan_voicemngr_get_sip_client_id(sub);
+ if (sip_client_id >= 0 && sip_client_id < MAX_SIP_CLIENTS) {
+ line_stats[sip_client_id].txpkts++;
+ line_stats[sip_client_id].txbytes += frame.datalen;
+ } else {
+ ast_debug(9, "Wrong sip client id: %d\n", sip_client_id);
+ }
+ // write header values into frame so asterisk uses the same RTP header as DSP
+ frame.seqno = (ntohl(packet_buf32[0]) & 0xffff);
+ frame.ts = ntohl(packet_buf32[1]);
+ frame.ssrc = ntohl(packet_buf32[2]);
+ } else if (packet_type == CHAN_VOICEMNGR_RTCP_SR || packet_type == CHAN_VOICEMNGR_RTCP_RR) {
+ frame.frametype = AST_FRAME_RTCP;
+ frame.data.ptr = ap->rtp;
+ frame.datalen = ap->rtp_size;
+ frame.subclass.integer = (packet_type == CHAN_VOICEMNGR_RTCP_SR ? RTCP_SR : RTCP_RR);
+ chan_voicemngr_process_outgoing_rtcp_packet(sub, ap->rtp, ap->rtp_size);
} else {
- ast_debug(9, "Wrong sip client id: %d\n", sip_client_id);
- }
- // write header values into frame so asterisk uses the same RTP header as DSP
- frame.seqno = (ntohl(packet_buf32[0]) & 0xffff);
- frame.ts = ntohl(packet_buf32[1]);
- frame.ssrc = ntohl(packet_buf32[2]);
- } else if (packet_type == CHAN_VOICEMNGR_RTCP_SR || packet_type == CHAN_VOICEMNGR_RTCP_RR) {
- frame.frametype = AST_FRAME_RTCP;
- frame.data.ptr = ap->rtp;
- frame.datalen = ap->rtp_size;
- frame.subclass.integer = (packet_type == CHAN_VOICEMNGR_RTCP_SR ? RTCP_SR : RTCP_RR);
- chan_voicemngr_process_outgoing_rtcp_packet(sub, ap->rtp, ap->rtp_size);
- } else {
- //ast_debug(5, "Dropping RTP frame of type %d.\n", packet_type);
- drop_frame=1;
- //pvt_unlock(sub->parent);
- }
- //ast_mutex_unlock(&sub->parent->lock);
- //pvt_unlock(sub->parent);
+ ast_debug(5, "Drop RTP frame of type %d.\n", packet_type);
+ }
- if (owner) {
- if (!drop_frame && (ast_channel_state(owner) == AST_STATE_UP || ast_channel_state(owner) == AST_STATE_RING)) {
- ast_queue_frame(owner, &frame);
+ if (packet_type != CHAN_VOICEMNGR_UNKNOWN && !drop_frame && sub->owner) {
+ ast_channel_ref(sub->owner);
+ ast_queue_frame(sub->owner, &frame);
+ ast_channel_unref(sub->owner);
}
- ast_channel_unref(owner);
}
}
@@ -5038,7 +5024,6 @@ static void ubus_call_answer_rtp_stats(struct ubus_request *req, int type, struc
static int endpt_get_rtp_stats(struct chan_voicemngr_subchannel *sub) {
int line = sub->parent->line_id;
- struct ubus_context *local_ctx;
struct blob_buf bb;
int ret;
@@ -5053,12 +5038,6 @@ static int endpt_get_rtp_stats(struct chan_voicemngr_subchannel *sub) {
return -1;
}
- local_ctx = ubus_connect(NULL);
- if (!local_ctx) {
- ast_log(LOG_ERROR, "%s(): ubus_connect() failed\n", __func__);
- return -1;
- }
-
memset(&bb, 0, sizeof(bb));
if(blob_buf_init(&bb, 0)) {
return -1;
@@ -5066,12 +5045,9 @@ static int endpt_get_rtp_stats(struct chan_voicemngr_subchannel *sub) {
blobmsg_add_u32(&bb, "line", line);
blobmsg_add_u8(&bb, "reset", 1); // always reset RTP stats after get them
-
ast_log(LOG_DEBUG, "thread %d: ubus call endpt rtp_stats \"{'line':%d,'reset':true}\"", ast_get_tid(), line);
- ret = ubus_invoke(local_ctx, endpt_id, "rtp_stats", bb.head, ubus_call_answer_rtp_stats, sub, 500);
-
+ ret = ubus_invoke(get_shared_context(__FUNCTION__), endpt_id, "rtp_stats", bb.head, ubus_call_answer_rtp_stats, sub, 500);
blob_buf_free(&bb);
- ast_ubus_free_context(local_ctx);
if (ret != UBUS_STATUS_OK) {
ast_log(LOG_DEBUG, "ubus_invoke for rtp_stats failed with return value %d\n", ret);
--
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