From 237b2001e3ce5728f5521d3ee5350d7706e769d0 Mon Sep 17 00:00:00 2001
From: Grzegorz Sluja <grzegorz.sluja@sigma.se>
Date: Fri, 13 Oct 2023 12:19:11 +0200
Subject: [PATCH] Use the same header for RTP/RTCP packets in DSP and Asterisk
- SSRC of RTCP and RTP should are synchronized
- source1->identifier is the SSRC of the peer stream
---
src/channels/chan_voicemngr.c | 26 +++++++++-----------------
src/channels/chan_voicemngr.h | 2 --
2 files changed, 9 insertions(+), 19 deletions(-)
diff --git a/src/channels/chan_voicemngr.c b/src/channels/chan_voicemngr.c
index f161446..4934f45 100644
--- a/src/channels/chan_voicemngr.c
+++ b/src/channels/chan_voicemngr.c
@@ -79,7 +79,7 @@ static int chan_voicemngr_in_conference(const struct chan_voicemngr_pvt *p);
static int cwtimeout_cb(const void *data);
static int call_waiting_cb(const void *data);
static int r4hanguptimeout_cb(const void *data);
-static void chan_voicemngr_generate_rtp_packet(struct chan_voicemngr_subchannel *p, uint8_t *packet_buf, int type, int seqno, unsigned int rtp_timestamp);
+static void chan_voicemngr_generate_rtp_packet(struct chan_voicemngr_subchannel *p, uint8_t *packet_buf, int type, int seqno, unsigned int rtp_timestamp, unsigned int ssrc);
static void chan_voicemngr_process_incoming_rtcp_packet(struct chan_voicemngr_subchannel *p, uint8_t *rtcp_frame, uint32_t rtcp_size);
static void chan_voicemngr_process_outgoing_rtcp_packet(struct chan_voicemngr_subchannel *p, uint8_t *rtcp_frame, uint32_t rtcp_size);
static int chan_voicemngr_mute_connection(struct chan_voicemngr_subchannel *p);
@@ -1747,7 +1747,7 @@ static int chan_voicemngr_write(struct ast_channel *ast, struct ast_frame *frame
pvt_lock(sub->parent, "TELCHAN write frame");
/* generate the rtp header */
- chan_voicemngr_generate_rtp_packet(sub, ap->rtp, RTP_PT_CN, frame->seqno, rtp_timestamp);
+ chan_voicemngr_generate_rtp_packet(sub, ap->rtp, RTP_PT_CN, frame->seqno, rtp_timestamp, frame->ssrc);
sip_client_id = chan_voicemngr_get_sip_client_id(sub);
if (sip_client_id >= 0 && sip_client_id < MAX_SIP_CLIENTS) {
@@ -1783,7 +1783,7 @@ static int chan_voicemngr_write(struct ast_channel *ast, struct ast_frame *frame
pvt_lock(sub->parent, "TELCHAN write frame");
/* generate the rtp header */
- chan_voicemngr_generate_rtp_packet(sub, ap->rtp, map_ast_codec_id_to_rtp(frame->subclass.format), frame->seqno, rtp_timestamp);
+ chan_voicemngr_generate_rtp_packet(sub, ap->rtp, map_ast_codec_id_to_rtp(frame->subclass.format), frame->seqno, rtp_timestamp, frame->ssrc);
sip_client_id = chan_voicemngr_get_sip_client_id(sub);
if (sip_client_id >= 0 && sip_client_id < MAX_SIP_CLIENTS) {
@@ -3199,7 +3199,10 @@ static void audio_packet_handler(pe_packet_t *p) {
} else {
ast_debug(9, "Wrong sip client id: %d\n", sip_client_id);
}
- sub->dsp_ssrc = packet_buf32[2]; // save the DSP's SSRC
+ // 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;
@@ -3739,8 +3742,6 @@ static struct chan_voicemngr_pvt *chan_voicemngr_allocate_pvt(void)
sub->channel_state = ONHOOK;
sub->time_stamp = 0;
sub->sequence_number = 0;
- sub->far_end_ssrc = 0;
- sub->dsp_ssrc = 0;
sub->codec = -1;
sub->parent = tmp;
sub->cw_timer_id = -1;
@@ -4011,8 +4012,6 @@ static void chan_voicemngr_show_subchannels(struct ast_cli_args *a, struct chan_
ast_cli(a->fd, " Connection init : %d\n", sub->connection_init);
ast_cli(a->fd, " Codec used : %s\n", chan_voicemngr_get_codec_string(sub->codec));
ast_cli(a->fd, " RTP sequence number : %d\n", sub->sequence_number);
- ast_cli(a->fd, " RTP Far End SSRC : %d\n", sub->far_end_ssrc);
- ast_cli(a->fd, " RTP DSP SSRC : %d\n", sub->dsp_ssrc);
ast_cli(a->fd, " RTP timestamp : %d\n", sub->time_stamp);
ast_cli(a->fd, " CW Timer id : %d\n", sub->cw_timer_id);
ast_cli(a->fd, " CW Beep Timer id : %d\n", sub->cw_tone_timer_id);
@@ -5358,9 +5357,6 @@ static int chan_voicemngr_signal_callerid(struct ast_channel *chan, struct chan_
static int chan_voicemngr_create_connection(struct chan_voicemngr_subchannel *sub) {
if (!sub->connection_init) {
- /* generate random nr for rtp header */
- sub->far_end_ssrc = rand();
-
ast_debug(1, "Creating virtual Asterisk connection for pvt line_id=%i connection_id=%d\n", sub->parent->line_id, sub->connection_id);
sub->connection_init = 1;
sub->remote_jitter_count = 0;
@@ -5616,7 +5612,7 @@ static int chan_voicemngr_close_connection(struct chan_voicemngr_subchannel *sub
}
/* Generate rtp payload, 12 bytes of header and 160 bytes of ulaw payload */
-static void chan_voicemngr_generate_rtp_packet(struct chan_voicemngr_subchannel *sub, uint8_t *packet_buf, int type, int seqno, unsigned int rtp_timestamp)
+static void chan_voicemngr_generate_rtp_packet(struct chan_voicemngr_subchannel *sub, uint8_t *packet_buf, int type, int seqno, unsigned int rtp_timestamp, unsigned int ssrc)
{
unsigned short* packet_buf16 = (unsigned short*)packet_buf;
unsigned int* packet_buf32 = (unsigned int*)packet_buf;
@@ -5630,7 +5626,7 @@ static void chan_voicemngr_generate_rtp_packet(struct chan_voicemngr_subchannel
packet_buf16[1] = htons(seqno ? seqno : sub->sequence_number++); //Add sequence number
packet_buf32[1] = htonl(rtp_timestamp ? rtp_timestamp : sub->time_stamp); //Add timestamp
sub->time_stamp += sub->period*8;
- packet_buf32[2] = sub->far_end_ssrc;
+ packet_buf32[2] = htonl(ssrc);
}
/*
@@ -5644,7 +5640,6 @@ static void chan_voicemngr_process_incoming_rtcp_packet(struct chan_voicemngr_su
{
struct rtcp_header_t *rtcp_hdr = (struct rtcp_header_t *)rtcp_frame;
uint8_t *packet_end = rtcp_frame + rtcp_size;
- unsigned int* packet_buf32 = (unsigned int*)rtcp_frame;
while ((uint8_t *)rtcp_hdr + sizeof(struct rtcp_header_t) <= packet_end && // Minimum RTCP packet size validation
RTCP_GET_VERSION(rtcp_hdr) == RTP_VERSION && // RTP version validation
@@ -5654,14 +5649,11 @@ static void chan_voicemngr_process_incoming_rtcp_packet(struct chan_voicemngr_su
p->remote_jitter_count++;
p->farEndInterarrivalJitter = RTCP_SR_GET_INTERARRIVAL_JITTER(rtcp_hdr);
p->totalFarEndInterarrivalJitter += p->farEndInterarrivalJitter;
- packet_buf32[7] = p->dsp_ssrc; // replace Source Identifier with dsp's SSRC
/* Intentional fall through */
case RTCP_RR:
case RTCP_SDES:
case RTCP_XR:
case RTCP_BYE:
- // Replace SSRC with far end SSRC for all types of RTCP packets above
- rtcp_hdr->ssrc = p->far_end_ssrc;
break;
default:
diff --git a/src/channels/chan_voicemngr.h b/src/channels/chan_voicemngr.h
index 8daab08..38343ab 100644
--- a/src/channels/chan_voicemngr.h
+++ b/src/channels/chan_voicemngr.h
@@ -152,8 +152,6 @@ struct chan_voicemngr_subchannel {
uint16_t sequence_number; /* Endpoint RTP sequence number state */
unsigned int time_stamp; /* Endpoint RTP time stamp state */
unsigned int period; /* Endpoint RTP period */
- unsigned int far_end_ssrc; /* Endpoint RTP synchronization source of far end */
- unsigned int dsp_ssrc; /* Endpoint RTP synchronization source of DUT's DSP */
int codec; /* Used codec */
struct chan_voicemngr_pvt *parent; /* chan_voicemngr_line owning this subchannel */
int cw_timer_id; /* Current call waiting timer id, -1 if no active timer */
--
GitLab