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FS-8315 #resolve [rtp_media_timeout not working]

This commit is contained in:
Anthony Minessale 2015-10-07 11:21:51 -05:00
parent 7f85faf21e
commit 3a24b972c0
2 changed files with 28 additions and 8 deletions
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4
src/switch_jitterbuffer.c
View File

@ -1138,6 +1138,10 @@ SWITCH_DECLARE(switch_status_t) switch_jb_get_packet(switch_jb_t *jb, switch_rtp
switch_mutex_lock(jb->mutex);
if (jb->complete_frames == 0) {
switch_goto_status(SWITCH_STATUS_BREAK, end);
}
if (jb->complete_frames < jb->frame_len) {
jb_debug(jb, 2, "BUFFERING %u/%u\n", jb->complete_frames , jb->frame_len);
switch_goto_status(SWITCH_STATUS_MORE_DATA, end);

32
src/switch_rtp.c
View File

@ -438,6 +438,8 @@ struct switch_rtp {
uint8_t has_rtp;
uint8_t has_rtcp;
uint8_t has_ice;
uint8_t punts;
uint8_t clean;
#ifdef ENABLE_ZRTP
zrtp_session_t *zrtp_session;
zrtp_profile_t *zrtp_profile;
@ -4060,20 +4062,22 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_activate_jitter_buffer(switch_rtp_t *
max_queue_frames = queue_frames * 3;
}
READ_INC(rtp_session);
if (rtp_session->jb) {
status = switch_jb_set_frames(rtp_session->jb, queue_frames, max_queue_frames);
} else {
READ_INC(rtp_session);
status = switch_jb_create(&rtp_session->jb, SJB_AUDIO, queue_frames, max_queue_frames, rtp_session->pool);
switch_jb_set_session(rtp_session->jb, rtp_session->session);
if (switch_true(switch_channel_get_variable_dup(switch_core_session_get_channel(rtp_session->session), "jb_use_timestamps", SWITCH_FALSE, -1))) {
switch_jb_ts_mode(rtp_session->jb, samples_per_packet, samples_per_second);
}
//switch_jb_debug_level(rtp_session->jb, 10);
READ_DEC(rtp_session);
}
READ_DEC(rtp_session);
return status;
}
@ -5022,6 +5026,12 @@ static switch_status_t read_rtp_packet(switch_rtp_t *rtp_session, switch_size_t
tries++;
if (tries > 20) {
if (rtp_session->jb && !rtp_session->pause_jb && jb_valid(rtp_session)) {
switch_jb_reset(rtp_session->jb);
}
rtp_session->punts++;
rtp_session->clean = 0;
*bytes = 0;
return SWITCH_STATUS_BREAK;
}
@ -5108,8 +5118,8 @@ static switch_status_t read_rtp_packet(switch_rtp_t *rtp_session, switch_size_t
}
}
rtp_session->missed_count = 0;
if (rtp_session->has_rtp) {
rtp_session->missed_count = 0;
switch_cp_addr(rtp_session->rtp_from_addr, rtp_session->from_addr);
rtp_session->last_rtp_hdr = rtp_session->recv_msg.header;
}
@ -5554,8 +5564,12 @@ static switch_status_t read_rtp_packet(switch_rtp_t *rtp_session, switch_size_t
switch(jstatus) {
case SWITCH_STATUS_MORE_DATA:
block = 1;
goto more;
if (rtp_session->punts < 4) {
block = 1;
goto more;
}
*bytes = 0;
break;
case SWITCH_STATUS_NOTFOUND:
{
int pt = get_recv_payload(rtp_session);
@ -5574,7 +5588,9 @@ static switch_status_t read_rtp_packet(switch_rtp_t *rtp_session, switch_size_t
rtp_session->stats.inbound.jb_packet_count++;
status = SWITCH_STATUS_SUCCESS;
rtp_session->last_rtp_hdr = rtp_session->recv_msg.header;
if (++rtp_session->clean > 200) {
rtp_session->punts = 0;
}
if (!xcheck_jitter) {
check_jitter(rtp_session);
xcheck_jitter = *bytes;
@ -6217,9 +6233,9 @@ static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_
ret = -1;
goto end;
}
if (rtp_session->max_missed_packets && read_loops == 1 && !rtp_session->flags[SWITCH_RTP_FLAG_VIDEO]) {
if (bytes) {
if (bytes && status == SWITCH_STATUS_SUCCESS) {
rtp_session->missed_count = 0;
} else if (++rtp_session->missed_count >= rtp_session->max_missed_packets) {
ret = -2;