diff options
Diffstat (limited to 'openbsc')
| -rw-r--r-- | openbsc/contrib/testconv/testconv_main.c | 52 | ||||
| -rw-r--r-- | openbsc/include/openbsc/mgcp.h | 5 | ||||
| -rw-r--r-- | openbsc/include/openbsc/mgcp_internal.h | 3 | ||||
| -rw-r--r-- | openbsc/src/libmgcp/mgcp_network.c | 30 | ||||
| -rw-r--r-- | openbsc/src/libmgcp/mgcp_protocol.c | 18 | ||||
| -rw-r--r-- | openbsc/src/libmgcp/mgcp_vty.c | 22 | ||||
| -rw-r--r-- | openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c | 229 |
7 files changed, 262 insertions, 97 deletions
diff --git a/openbsc/contrib/testconv/testconv_main.c b/openbsc/contrib/testconv/testconv_main.c index c2785f23e..aee73048c 100644 --- a/openbsc/contrib/testconv/testconv_main.c +++ b/openbsc/contrib/testconv/testconv_main.c @@ -38,10 +38,10 @@ int mgcp_get_trans_frame_size(void *state_, int nsamples, int dst); int main(int argc, char **argv) { - char buf[4096] = {0}; + char buf[4096] = {0x80, 0}; int cc, rc; - struct mgcp_rtp_end dst_end = {0}; - struct mgcp_rtp_end src_end = {0}; + struct mgcp_rtp_end *dst_end; + struct mgcp_rtp_end *src_end; struct mgcp_trunk_config tcfg = {{0}}; struct mgcp_endpoint endp = {0}; struct mgcp_process_rtp_state *state; @@ -52,39 +52,63 @@ int main(int argc, char **argv) tcfg.endpoints = &endp; tcfg.number_endpoints = 1; endp.tcfg = &tcfg; + mgcp_free_endp(&endp); + + dst_end = &endp.bts_end; + src_end = &endp.net_end; if (argc <= 2) errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16}"); - if ((src_end.payload_type = audio_name_to_type(argv[1])) == -1) + if ((src_end->payload_type = audio_name_to_type(argv[1])) == -1) errx(1, "invalid input format '%s'", argv[1]); - if ((dst_end.payload_type = audio_name_to_type(argv[2])) == -1) + if ((dst_end->payload_type = audio_name_to_type(argv[2])) == -1) errx(1, "invalid output format '%s'", argv[2]); - rc = mgcp_transcoding_setup(&endp, &dst_end, &src_end); + rc = mgcp_transcoding_setup(&endp, dst_end, src_end); if (rc < 0) errx(1, "setup failed: %s", strerror(-rc)); - state = dst_end.rtp_process_data; + state = dst_end->rtp_process_data; OSMO_ASSERT(state != NULL); in_size = mgcp_transcoding_get_frame_size(state, 160, 0); OSMO_ASSERT(sizeof(buf) >= in_size + 12); + buf[1] = src_end->payload_type; + *(uint16_t*)(buf+2) = htons(1); + *(uint32_t*)(buf+4) = htonl(0); + *(uint32_t*)(buf+8) = htonl(0xaabbccdd); + while ((cc = read(0, buf + 12, in_size))) { + int cont; + int len; + if (cc != in_size) err(1, "read"); cc += 12; /* include RTP header */ - rc = mgcp_transcoding_process_rtp(&endp, &dst_end, - buf, &cc, sizeof(buf)); - if (rc < 0) - errx(1, "processing failed: %s", strerror(-rc)); + len = cc; + + do { + cont = mgcp_transcoding_process_rtp(&endp, dst_end, + buf, &len, sizeof(buf)); + if (cont == -EAGAIN) { + fprintf(stderr, "Got EAGAIN\n"); + break; + } + + if (cont < 0) + errx(1, "processing failed: %s", strerror(-cont)); + + len -= 12; /* ignore RTP header */ + + if (write(1, buf + 12, len) != len) + err(1, "write"); - cc -= 12; /* ignore RTP header */ - if (write(1, buf + 12, cc) != cc) - err(1, "write"); + len = cont; + } while (len > 0); } return 0; } diff --git a/openbsc/include/openbsc/mgcp.h b/openbsc/include/openbsc/mgcp.h index 002dd7c6e..d4d614099 100644 --- a/openbsc/include/openbsc/mgcp.h +++ b/openbsc/include/openbsc/mgcp.h @@ -87,7 +87,8 @@ typedef int (*mgcp_policy)(struct mgcp_trunk_config *cfg, int endpoint, int stat typedef int (*mgcp_reset)(struct mgcp_trunk_config *cfg); typedef int (*mgcp_rqnt)(struct mgcp_endpoint *endp, char tone); -typedef int (*mgcp_processing)(struct mgcp_rtp_end *dst_end, +typedef int (*mgcp_processing)(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, char *data, int *len, int buf_size); typedef int (*mgcp_processing_setup)(struct mgcp_endpoint *endp, struct mgcp_rtp_end *dst_end, @@ -181,6 +182,8 @@ struct mgcp_config { struct mgcp_port_range transcoder_ports; int endp_dscp; + int bts_force_ptime; + mgcp_change change_cb; mgcp_policy policy_cb; mgcp_reset reset_cb; diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h index ac136a377..9f0c0f9e2 100644 --- a/openbsc/include/openbsc/mgcp_internal.h +++ b/openbsc/include/openbsc/mgcp_internal.h @@ -89,6 +89,7 @@ struct mgcp_rtp_end { char *audio_name; char *subtype_name; int output_enabled; + int force_output_ptime; /* RTP patching */ int force_constant_ssrc; /* -1: always, 0: don't, 1: once */ @@ -210,7 +211,7 @@ void mgcp_state_calc_loss(struct mgcp_rtp_state *s, struct mgcp_rtp_end *, uint32_t mgcp_state_calc_jitter(struct mgcp_rtp_state *); /* payload processing default functions */ -int mgcp_rtp_processing_default(struct mgcp_rtp_end *dst_end, +int mgcp_rtp_processing_default(struct mgcp_endpoint *endp, struct mgcp_rtp_end *dst_end, char *data, int *len, int buf_size); int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp, diff --git a/openbsc/src/libmgcp/mgcp_network.c b/openbsc/src/libmgcp/mgcp_network.c index 05c3e7749..219d3f990 100644 --- a/openbsc/src/libmgcp/mgcp_network.c +++ b/openbsc/src/libmgcp/mgcp_network.c @@ -340,7 +340,7 @@ static int align_rtp_timestamp_offset(struct mgcp_endpoint *endp, return timestamp_error; } -int mgcp_rtp_processing_default(struct mgcp_rtp_end *dst_end, +int mgcp_rtp_processing_default(struct mgcp_endpoint *endp, struct mgcp_rtp_end *dst_end, char *data, int *len, int buf_size) { return 0; @@ -614,12 +614,28 @@ int mgcp_send(struct mgcp_endpoint *endp, int dest, int is_rtp, if (!rtp_end->output_enabled) rtp_end->dropped_packets += 1; else if (is_rtp) { - mgcp_patch_and_count(endp, rtp_state, rtp_end, addr, buf, rc); - endp->cfg->rtp_processing_cb(rtp_end, buf, &rc, RTP_BUF_SIZE); - forward_data(rtp_end->rtp.fd, &endp->taps[tap_idx], buf, rc); - return mgcp_udp_send(rtp_end->rtp.fd, - &rtp_end->addr, - rtp_end->rtp_port, buf, rc); + int cont; + int nbytes = 0; + int len = rc; + mgcp_patch_and_count(endp, rtp_state, rtp_end, addr, buf, len); + do { + cont = endp->cfg->rtp_processing_cb(endp, rtp_end, + buf, &len, RTP_BUF_SIZE); + if (cont < 0) + break; + + forward_data(rtp_end->rtp.fd, &endp->taps[tap_idx], + buf, len); + rc = mgcp_udp_send(rtp_end->rtp.fd, + &rtp_end->addr, + rtp_end->rtp_port, buf, len); + + if (rc <= 0) + return rc; + nbytes += rc; + len = cont; + } while (len > 0); + return nbytes; } else if (!tcfg->omit_rtcp) { return mgcp_udp_send(rtp_end->rtcp.fd, &rtp_end->addr, diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c index 6e974f147..21b9ff0f8 100644 --- a/openbsc/src/libmgcp/mgcp_protocol.c +++ b/openbsc/src/libmgcp/mgcp_protocol.c @@ -621,6 +621,15 @@ static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp, rtp->channels = channels; rtp->subtype_name = talloc_strdup(ctx, audio_codec); rtp->audio_name = talloc_strdup(ctx, audio_name); + + if (!strcmp(audio_codec, "G729")) { + rtp->frame_duration_num = 10; + rtp->frame_duration_den = 1000; + } else { + rtp->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM; + rtp->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN; + } + if (channels != 1) LOGP(DMGCP, LOGL_NOTICE, "Channels != 1 in SDP: '%s'\n", audio_name); @@ -944,11 +953,16 @@ mgcp_header_done: set_audio_info(p->cfg, &endp->bts_end, tcfg->audio_payload, tcfg->audio_name); endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints, tcfg->audio_fmtp_extra); - if (have_sdp) { + if (have_sdp) parse_sdp_data(&endp->net_end, p); - setup_rtp_processing(endp); + + if (p->cfg->bts_force_ptime) { + endp->bts_end.packet_duration_ms = p->cfg->bts_force_ptime; + endp->bts_end.force_output_ptime = 1; } + setup_rtp_processing(endp); + /* policy CB */ if (p->cfg->policy_cb) { int rc; diff --git a/openbsc/src/libmgcp/mgcp_vty.c b/openbsc/src/libmgcp/mgcp_vty.c index 1f8a63ab7..26b570627 100644 --- a/openbsc/src/libmgcp/mgcp_vty.c +++ b/openbsc/src/libmgcp/mgcp_vty.c @@ -366,6 +366,26 @@ ALIAS_DEPRECATED(cfg_mgcp_rtp_ip_dscp, cfg_mgcp_rtp_ip_tos_cmd, RTP_STR "Apply IP_TOS to the audio stream\n" "The DSCP value\n") +#define FORCE_PTIME_STR "Force a fixed ptime for packets sent to the BTS" +DEFUN(cfg_mgcp_rtp_force_ptime, + cfg_mgcp_rtp_force_ptime_cmd, + "rtp force-ptime (10|20|40)", + RTP_STR FORCE_PTIME_STR + "The required ptime (packet duration) in ms\n") +{ + g_cfg->bts_force_ptime = atoi(argv[0]); + return CMD_SUCCESS; +} + +DEFUN(cfg_mgcp_no_rtp_force_ptime, + cfg_mgcp_no_rtp_force_ptime_cmd, + "no rtp force-ptime", + NO_STR RTP_STR FORCE_PTIME_STR) +{ + g_cfg->bts_force_ptime = 0; + return CMD_SUCCESS; +} + DEFUN(cfg_mgcp_sdp_fmtp_extra, cfg_mgcp_sdp_fmtp_extra_cmd, "sdp audio fmtp-extra .NAME", @@ -1123,6 +1143,8 @@ int mgcp_vty_init(void) install_element(MGCP_NODE, &cfg_mgcp_rtp_transcoder_base_cmd); install_element(MGCP_NODE, &cfg_mgcp_rtp_ip_dscp_cmd); install_element(MGCP_NODE, &cfg_mgcp_rtp_ip_tos_cmd); + install_element(MGCP_NODE, &cfg_mgcp_rtp_force_ptime_cmd); + install_element(MGCP_NODE, &cfg_mgcp_no_rtp_force_ptime_cmd); install_element(MGCP_NODE, &cfg_mgcp_rtp_keepalive_cmd); install_element(MGCP_NODE, &cfg_mgcp_rtp_keepalive_once_cmd); install_element(MGCP_NODE, &cfg_mgcp_no_rtp_keepalive_cmd); diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c index 91c0c383c..581cd3293 100644 --- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c +++ b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c @@ -1,5 +1,4 @@ /* - * (C) 2014 by Sysmocom s.f.m.c. GmbH * (C) 2014 by On-Waves * All Rights Reserved * @@ -22,7 +21,8 @@ #include <string.h> #include <errno.h> -#include "bscconfig.h" + +#include "../../bscconfig.h" #include "g711common.h" #include <gsm.h> @@ -70,6 +70,14 @@ struct mgcp_process_rtp_state { } dst; size_t dst_frame_size; size_t dst_samples_per_frame; + int dst_packet_duration; + + int is_running; + uint16_t next_seq; + uint32_t next_time; + int16_t samples[10*160]; + size_t sample_cnt; + size_t sample_offs; }; int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst) @@ -302,6 +310,9 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp, break; } + if (dst_end->force_output_ptime) + state->dst_packet_duration = mgcp_rtp_packet_duration(endp, dst_end); + LOGP(DMGCP, LOGL_INFO, "Initialized RTP processing on: 0x%x " "conv: %d (%d, %d, %s) -> %d (%d, %d, %s)\n", @@ -330,44 +341,21 @@ void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp, *audio_name = endp->net_end.audio_name; } - -int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp, - struct mgcp_rtp_end *dst_end, - char *data, int *len, int buf_size) +static int decode_audio(struct mgcp_process_rtp_state *state, + uint8_t **src, size_t *nbytes) { - struct mgcp_process_rtp_state *state = dst_end->rtp_process_data; - size_t rtp_hdr_size = 12; - char *payload_data = data + rtp_hdr_size; - int payload_len = *len - rtp_hdr_size; - size_t sample_cnt = 0; - size_t sample_idx; - int16_t samples[10*160]; - uint8_t *src = (uint8_t *)payload_data; - uint8_t *dst = (uint8_t *)payload_data; - size_t nbytes = payload_len; - size_t frame_remainder; - - if (!state) - return 0; - - if (state->src_fmt == state->dst_fmt) - return 0; - - /* TODO: check payload type (-> G.711 comfort noise) */ - - /* Decode src into samples */ - while (nbytes >= state->src_frame_size) { - if (sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(samples)) { + while (*nbytes >= state->src_frame_size) { + if (state->sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(state->samples)) { LOGP(DMGCP, LOGL_ERROR, "Sample buffer too small: %d > %d.\n", - sample_cnt + state->src_samples_per_frame, - ARRAY_SIZE(samples)); + state->sample_cnt + state->src_samples_per_frame, + ARRAY_SIZE(state->samples)); return -ENOSPC; } switch (state->src_fmt) { case AF_GSM: if (gsm_decode(state->src.gsm_handle, - (gsm_byte *)src, samples + sample_cnt) < 0) { + (gsm_byte *)*src, state->samples + state->sample_cnt) < 0) { LOGP(DMGCP, LOGL_ERROR, "Failed to decode GSM.\n"); return -EINVAL; @@ -375,54 +363,44 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp, break; #ifdef HAVE_BCG729 case AF_G729: - bcg729Decoder(state->src.g729_dec, src, 0, samples + sample_cnt); + bcg729Decoder(state->src.g729_dec, *src, 0, state->samples + state->sample_cnt); break; #endif case AF_PCMA: - alaw_decode(src, samples + sample_cnt, + alaw_decode(*src, state->samples + state->sample_cnt, state->src_samples_per_frame); break; case AF_S16: - memmove(samples + sample_cnt, src, + memmove(state->samples + state->sample_cnt, *src, state->src_frame_size); break; case AF_L16: - l16_decode(src, samples + sample_cnt, + l16_decode(*src, state->samples + state->sample_cnt, state->src_samples_per_frame); break; default: break; } - src += state->src_frame_size; - nbytes -= state->src_frame_size; - sample_cnt += state->src_samples_per_frame; - } - - /* Add silence if necessary */ - frame_remainder = sample_cnt % state->dst_samples_per_frame; - if (frame_remainder) { - size_t silence = state->dst_samples_per_frame - frame_remainder; - if (sample_cnt + silence > ARRAY_SIZE(samples)) { - LOGP(DMGCP, LOGL_ERROR, - "Sample buffer too small for silence: %d > %d.\n", - sample_cnt + silence, - ARRAY_SIZE(samples)); - return -ENOSPC; - } - - while (silence > 0) { - samples[sample_cnt] = 0; - sample_cnt += 1; - silence -= 1; - } + *src += state->src_frame_size; + *nbytes -= state->src_frame_size; + state->sample_cnt += state->src_samples_per_frame; } + return 0; +} +static int encode_audio(struct mgcp_process_rtp_state *state, + uint8_t *dst, size_t buf_size, size_t max_samples) +{ + int nbytes = 0; + size_t nsamples = 0; /* Encode samples into dst */ - sample_idx = 0; - nbytes = 0; - while (sample_idx + state->dst_samples_per_frame <= sample_cnt) { + while (nsamples + state->dst_samples_per_frame <= max_samples) { if (nbytes + state->dst_frame_size > buf_size) { - LOGP(DMGCP, LOGL_ERROR, + if (nbytes > 0) + break; + + /* Not even one frame fits into the buffer */ + LOGP(DMGCP, LOGL_INFO, "Encoding (RTP) buffer too small: %d > %d.\n", nbytes + state->dst_frame_size, buf_size); return -ENOSPC; @@ -430,23 +408,24 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp, switch (state->dst_fmt) { case AF_GSM: gsm_encode(state->dst.gsm_handle, - samples + sample_idx, dst); + state->samples + state->sample_offs, dst); break; #ifdef HAVE_BCG729 case AF_G729: bcg729Encoder(state->dst.g729_enc, - samples + sample_idx, dst); + state->samples + state->sample_offs, dst); break; #endif case AF_PCMA: - alaw_encode(samples + sample_idx, dst, + alaw_encode(state->samples + state->sample_offs, dst, state->src_samples_per_frame); break; case AF_S16: - memmove(dst, samples + sample_idx, state->dst_frame_size); + memmove(dst, state->samples + state->sample_offs, + state->dst_frame_size); break; case AF_L16: - l16_encode(samples + sample_idx, dst, + l16_encode(state->samples + state->sample_offs, dst, state->src_samples_per_frame); break; default: @@ -454,12 +433,118 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp, } dst += state->dst_frame_size; nbytes += state->dst_frame_size; - sample_idx += state->dst_samples_per_frame; + state->sample_offs += state->dst_samples_per_frame; + nsamples += state->dst_samples_per_frame; } + state->sample_cnt -= nsamples; + return nbytes; +} - *len = rtp_hdr_size + nbytes; - /* Patch payload type */ - data[1] = (data[1] & 0x80) | (dst_end->payload_type & 0x7f); +int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, + char *data, int *len, int buf_size) +{ + struct mgcp_process_rtp_state *state = dst_end->rtp_process_data; + size_t rtp_hdr_size = 12; + char *payload_data = data + rtp_hdr_size; + int payload_len = *len - rtp_hdr_size; + uint8_t *src = (uint8_t *)payload_data; + uint8_t *dst = (uint8_t *)payload_data; + size_t nbytes = payload_len; + size_t nsamples; + size_t max_samples; + uint32_t ts_no; + int rc; - return 0; + if (!state) + return 0; + + if (state->src_fmt == state->dst_fmt) { + if (!state->dst_packet_duration) + return 0; + + /* TODO: repackage without transcoding */ + } + + /* If the remaining samples do not fit into a fixed ptime, + * a) discard them, if the next packet is much later + * b) add silence and * send it, if the current packet is not + * yet too late + * c) append the sample data, if the timestamp matches exactly + */ + + /* TODO: check payload type (-> G.711 comfort noise) */ + + if (payload_len > 0) { + ts_no = ntohl(*(uint32_t*)(data+4)); + if (!state->is_running) + state->next_seq = ntohs(*(uint32_t*)(data+4)); + + state->is_running = 1; + + if (state->sample_cnt > 0) { + int32_t delta = ts_no - state->next_time; + /* TODO: check sequence? reordering? packet loss? */ + + if (delta > state->sample_cnt) + /* There is a time gap between the last packet + * and the current one. Just discard the + * partial data that is left in the buffer. + * TODO: This can be improved by adding silence + * instead if the delta is small enough. + */ + state->sample_cnt = 0; + else if (delta < 0) { + LOGP(DMGCP, LOGL_NOTICE, + "RTP time jumps backwards, delta = %d, " + "discarding buffered samples\n", + delta); + state->sample_cnt = 0; + state->sample_offs = 0; + return -EAGAIN; + } + + /* Make sure the samples start without offset */ + if (state->sample_offs && state->sample_cnt) + memmove(&state->samples[0], + &state->samples[state->sample_offs], + state->sample_cnt * + sizeof(state->samples[0])); + } + + state->sample_offs = 0; + + /* Append decoded audio to samples */ + decode_audio(state, &src, &nbytes); + + if (nbytes > 0) + LOGP(DMGCP, LOGL_NOTICE, + "Skipped audio frame in RTP packet: %d octets\n", + nbytes); + } else + ts_no = state->next_time; + + if (state->sample_cnt < state->dst_packet_duration) + return -EAGAIN; + + max_samples = + state->dst_packet_duration ? + state->dst_packet_duration : state->sample_cnt; + + nsamples = state->sample_cnt; + + rc = encode_audio(state, dst, buf_size, max_samples); + if (rc <= 0) + return rc; + + nsamples -= state->sample_cnt; + + *len = rtp_hdr_size + rc; + *(uint16_t*)(data+2) = htonl(state->next_seq); + *(uint32_t*)(data+4) = htonl(ts_no); + + state->next_seq += 1; + state->next_time = ts_no + nsamples; + + return nsamples ? rtp_hdr_size : 0; } |