diff options
-rw-r--r-- | include/asterisk/rtp_engine.h | 6 | ||||
-rw-r--r-- | include/asterisk/utils.h | 13 | ||||
-rw-r--r-- | res/res_musiconhold.c | 1 | ||||
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 3 | ||||
-rw-r--r-- | res/res_rtp_asterisk.c | 171 |
5 files changed, 183 insertions, 11 deletions
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h index 8f044ce17..3426b2a1e 100644 --- a/include/asterisk/rtp_engine.h +++ b/include/asterisk/rtp_engine.h @@ -292,10 +292,14 @@ struct ast_rtp_payload_type { #define AST_RTP_RTCP_SR 200 /*! Receiver Report */ #define AST_RTP_RTCP_RR 201 +/*! Transport Layer Feed Back (From RFC4585 also RFC5104) */ +#define AST_RTP_RTCP_RTPFB 205 /*! Payload Specific Feed Back (From RFC4585 also RFC5104) */ -#define AST_RTP_RTCP_PSFB 206 +#define AST_RTP_RTCP_PSFB 206 /* Common RTCP feedback message types */ +/*! Generic NACK (From RFC4585 also RFC5104) */ +#define AST_RTP_RTCP_FMT_NACK 1 /*! Picture loss indication (From RFC4585) */ #define AST_RTP_RTCP_FMT_PLI 1 /*! Full INTRA-frame Request (From RFC5104) */ diff --git a/include/asterisk/utils.h b/include/asterisk/utils.h index 4da7fa465..b892cda9e 100644 --- a/include/asterisk/utils.h +++ b/include/asterisk/utils.h @@ -578,6 +578,13 @@ void DO_CRASH_NORETURN __ast_assert_failed(int condition, const char *condition_ #ifdef AST_DEVMODE #define ast_assert(a) _ast_assert(a, # a, __FILE__, __LINE__, __PRETTY_FUNCTION__) +#define ast_assert_return(a, ...) \ +({ \ + if (__builtin_expect(!(a), 1)) { \ + _ast_assert(0, # a, __FILE__, __LINE__, __PRETTY_FUNCTION__); \ + return __VA_ARGS__; \ + }\ +}) static void force_inline _ast_assert(int condition, const char *condition_str, const char *file, int line, const char *function) { if (__builtin_expect(!condition, 1)) { @@ -586,6 +593,12 @@ static void force_inline _ast_assert(int condition, const char *condition_str, c } #else #define ast_assert(a) +#define ast_assert_return(a, ...) \ +({ \ + if (__builtin_expect(!(a), 1)) { \ + return __VA_ARGS__; \ + }\ +}) #endif /*! diff --git a/res/res_musiconhold.c b/res/res_musiconhold.c index 55b14c934..1c8728cf7 100644 --- a/res/res_musiconhold.c +++ b/res/res_musiconhold.c @@ -333,7 +333,6 @@ static int ast_moh_files_next(struct ast_channel *chan) } } else { state->announcement = 0; - state->samples = 0; } if (!state->class->total_files) { diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 03d37652f..14ed3b186 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -1096,6 +1096,9 @@ static void add_rtcp_fb_to_stream(struct ast_sip_session *session, attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* goog-remb")); pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr); + + attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* nack")); + pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr); } /*! \brief Function which negotiates an incoming media stream */ diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index c88903777..4ac20d551 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -71,6 +71,7 @@ #include "asterisk/smoother.h" #include "asterisk/uuid.h" #include "asterisk/test.h" +#include "asterisk/data_buffer.h" #ifdef HAVE_PJPROJECT #include "asterisk/res_pjproject.h" #endif @@ -92,6 +93,8 @@ #define TURN_STATE_WAIT_TIME 2000 +#define DEFAULT_RTP_BUFFER_SIZE 250 + /*! Full INTRA-frame Request / Fast Update Request (From RFC2032) */ #define RTCP_PT_FUR 192 /*! Sender Report (From RFC3550) */ @@ -373,6 +376,8 @@ struct ast_rtp { struct rtp_red *red; + struct ast_data_buffer *send_buffer; /*!< Buffer for storing sent packets for retransmission */ + #ifdef HAVE_PJPROJECT ast_cond_t cond; /*!< ICE/TURN condition for signaling */ @@ -509,6 +514,12 @@ struct rtp_red { long int prev_ts; }; +/*! \brief Structure for storing RTP packets for retransmission */ +struct ast_rtp_rtcp_nack_payload { + size_t size; /*!< The size of the payload */ + unsigned char buf[0]; /*!< The payload data */ +}; + AST_LIST_HEAD_NOLOCK(frame_list, ast_frame); /* Forward Declarations */ @@ -3675,6 +3686,11 @@ static int ast_rtp_destroy(struct ast_rtp_instance *instance) rtp->red = NULL; } + /* Destroy the send buffer if it was being used */ + if (rtp->send_buffer) { + ast_data_buffer_free(rtp->send_buffer); + } + ao2_cleanup(rtp->lasttxformat); ao2_cleanup(rtp->lastrxformat); ao2_cleanup(rtp->f.subclass.format); @@ -4369,7 +4385,7 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr } else { /* This is the first frame with sequence number we've seen, so start keeping track */ rtp->expectedseqno = frame->seqno + 1; - } + } } else { rtp->expectedseqno = -1; } @@ -4383,13 +4399,27 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr /* If we know the remote address construct a packet and send it out */ if (!ast_sockaddr_isnull(&remote_address)) { int hdrlen = 12, res, ice; + int packet_len = frame->datalen + hdrlen; unsigned char *rtpheader = (unsigned char *)(frame->data.ptr - hdrlen); put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (seqno) | (mark << 23))); put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts)); put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc)); - if ((res = rtp_sendto(instance, (void *)rtpheader, frame->datalen + hdrlen, 0, &remote_address, &ice)) < 0) { + /* If retransmissions are enabled, we need to store this packet for future use */ + if (rtp->send_buffer) { + struct ast_rtp_rtcp_nack_payload *payload; + + payload = ast_malloc(sizeof(*payload) + packet_len); + if (payload) { + payload->size = packet_len; + memcpy(payload->buf, rtpheader, packet_len); + ast_data_buffer_put(rtp->send_buffer, rtp->seqno, payload); + } + } + + res = rtp_sendto(instance, (void *)rtpheader, packet_len, 0, &remote_address, &ice); + if (res < 0) { if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) || (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) { ast_debug(1, "RTP Transmission error of packet %d to %s: %s\n", rtp->seqno, @@ -5145,8 +5175,8 @@ static void update_lost_stats(struct ast_rtp *rtp, unsigned int lost_packets) } /*! \pre instance is locked */ -static struct ast_rtp_instance *rtp_find_instance_by_ssrc(struct ast_rtp_instance *instance, - struct ast_rtp *rtp, unsigned int ssrc) +static struct ast_rtp_instance *__rtp_find_instance_by_ssrc(struct ast_rtp_instance *instance, + struct ast_rtp *rtp, unsigned int ssrc, int source) { int index; @@ -5158,8 +5188,9 @@ static struct ast_rtp_instance *rtp_find_instance_by_ssrc(struct ast_rtp_instanc /* Find the bundled child instance */ for (index = 0; index < AST_VECTOR_SIZE(&rtp->ssrc_mapping); ++index) { struct rtp_ssrc_mapping *mapping = AST_VECTOR_GET_ADDR(&rtp->ssrc_mapping, index); + unsigned int mapping_ssrc = source ? ast_rtp_get_ssrc(mapping->instance) : mapping->ssrc; - if (mapping->ssrc_valid && mapping->ssrc == ssrc) { + if (mapping->ssrc_valid && mapping_ssrc == ssrc) { return mapping->instance; } } @@ -5171,6 +5202,20 @@ static struct ast_rtp_instance *rtp_find_instance_by_ssrc(struct ast_rtp_instanc return NULL; } +/*! \pre instance is locked */ +static struct ast_rtp_instance *rtp_find_instance_by_packet_source_ssrc(struct ast_rtp_instance *instance, + struct ast_rtp *rtp, unsigned int ssrc) +{ + return __rtp_find_instance_by_ssrc(instance, rtp, ssrc, 0); +} + +/*! \pre instance is locked */ +static struct ast_rtp_instance *rtp_find_instance_by_media_source_ssrc(struct ast_rtp_instance *instance, + struct ast_rtp *rtp, unsigned int ssrc) +{ + return __rtp_find_instance_by_ssrc(instance, rtp, ssrc, 1); +} + static const char *rtcp_payload_type2str(unsigned int pt) { const char *str; @@ -5203,6 +5248,69 @@ static const char *rtcp_payload_type2str(unsigned int pt) return str; } +/*! \pre instance is locked */ +static int ast_rtp_rtcp_handle_nack(struct ast_rtp_instance *instance, unsigned int *nackdata, unsigned int position, + unsigned int length) +{ + struct ast_rtp *rtp = ast_rtp_instance_get_data(instance); + int res = 0; + int blp_index; + int packet_index; + int ice; + struct ast_rtp_rtcp_nack_payload *payload; + unsigned int current_word; + unsigned int pid; /* Packet ID which refers to seqno of lost packet */ + unsigned int blp; /* Bitmask of following lost packets */ + struct ast_sockaddr remote_address = { {0,} }; + + if (!rtp->send_buffer) { + ast_debug(1, "Tried to handle NACK request, but we don't have a RTP packet storage!\n"); + return res; + } + + ast_rtp_instance_get_remote_address(instance, &remote_address); + + /* + * We use index 3 because with feedback messages, the FCI (Feedback Control Information) + * does not begin until after the version, packet SSRC, and media SSRC words. + */ + for (packet_index = 3; packet_index < length; packet_index++) { + current_word = ntohl(nackdata[position + packet_index]); + pid = current_word >> 16; + /* We know the remote end is missing this packet. Go ahead and send it if we still have it. */ + payload = (struct ast_rtp_rtcp_nack_payload *)ast_data_buffer_get(rtp->send_buffer, pid); + if (payload) { + res += rtp_sendto(instance, payload->buf, payload->size, 0, &remote_address, &ice); + } else { + ast_debug(1, "Received NACK request for RTP packet with seqno %d, but we don't have it\n", pid); + } + /* + * The bitmask. Denoting the least significant bit as 1 and its most significant bit + * as 16, then bit i of the bitmask is set to 1 if the receiver has not received RTP + * packet (pid+i)(modulo 2^16). Otherwise, it is set to 0. We cannot assume bits set + * to 0 after a bit set to 1 have actually been received. + */ + blp = current_word & 0xFF; + blp_index = 1; + while (blp) { + if (blp & 1) { + /* Packet (pid + i)(modulo 2^16) is missing too. */ + unsigned int seqno = (pid + blp_index) % 65536; + payload = (struct ast_rtp_rtcp_nack_payload *)ast_data_buffer_get(rtp->send_buffer, seqno); + if (payload) { + res += rtp_sendto(instance, payload->buf, payload->size, 0, &remote_address, &ice); + } else { + ast_debug(1, "Remote end also requested RTP packet with seqno %d, but we don't have it\n", seqno); + } + } + blp >>= 1; + blp_index++; + } + } + + return res; +} + /* * Unshifted RTCP header bit field masks */ @@ -5243,6 +5351,7 @@ static const char *rtcp_payload_type2str(unsigned int pt) #define RTCP_RR_BLOCK_WORD_LENGTH 6 #define RTCP_HEADER_SSRC_LENGTH 2 #define RTCP_FB_REMB_BLOCK_WORD_LENGTH 4 +#define RTCP_FB_NACK_BLOCK_WORD_LENGTH 2 static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, const unsigned char *rtcpdata, size_t size, struct ast_sockaddr *addr) { @@ -5319,6 +5428,8 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c unsigned int ssrc_valid; unsigned int length; unsigned int min_length; + /*! Always use packet source SSRC to find the rtp instance unless explicitly told not to. */ + unsigned int use_packet_source = 1; struct ast_json *message_blob; RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup); @@ -5341,9 +5452,20 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c /* fall through */ case RTCP_PT_RR: min_length += (rc * RTCP_RR_BLOCK_WORD_LENGTH); + use_packet_source = 0; break; case RTCP_PT_FUR: break; + case AST_RTP_RTCP_RTPFB: + switch (rc) { + case AST_RTP_RTCP_FMT_NACK: + min_length += RTCP_FB_NACK_BLOCK_WORD_LENGTH; + break; + default: + break; + } + use_packet_source = 0; + break; case RTCP_PT_PSFB: switch (rc) { case AST_RTP_RTCP_FMT_REMB: @@ -5392,13 +5514,16 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c } rtcp_report->reception_report_count = rc; - ssrc = ntohl(rtcpheader[i + 1]); + ssrc = ntohl(rtcpheader[i + 2]); rtcp_report->ssrc = ssrc; break; case RTCP_PT_FUR: case RTCP_PT_PSFB: ssrc = ntohl(rtcpheader[i + 1]); break; + case AST_RTP_RTCP_RTPFB: + ssrc = ntohl(rtcpheader[i + 2]); + break; case RTCP_PT_SDES: case RTCP_PT_BYE: default: @@ -5417,7 +5542,15 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c /* Determine the appropriate instance for this */ if (ssrc_valid) { - child = rtp_find_instance_by_ssrc(transport, transport_rtp, ssrc); + /* + * Depending on the payload type, either the packet source or media source + * SSRC is used. + */ + if (use_packet_source) { + child = rtp_find_instance_by_packet_source_ssrc(transport, transport_rtp, ssrc); + } else { + child = rtp_find_instance_by_media_source_ssrc(transport, transport_rtp, ssrc); + } if (child && child != transport) { /* * It is safe to hold the child lock while holding the parent lock. @@ -5438,7 +5571,7 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c } if (ssrc_valid && rtp->themssrc_valid) { - if (ssrc != rtp->themssrc) { + if (ssrc != rtp->themssrc && use_packet_source) { /* * Skip over this RTCP record as it does not contain the * correct SSRC. We should not act upon RTCP records @@ -5581,6 +5714,24 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c transport_rtp->f.src = "RTP"; f = &transport_rtp->f; break; + case AST_RTP_RTCP_RTPFB: + switch (rc) { + case AST_RTP_RTCP_FMT_NACK: + /* If retransmissions are not enabled ignore this message */ + if (!rtp->send_buffer) { + break; + } + + if (rtcp_debug_test_addr(addr)) { + ast_verbose("Received generic RTCP NACK message\n"); + } + + ast_rtp_rtcp_handle_nack(instance, rtcpheader, position, length); + break; + default: + break; + } + break; case RTCP_PT_FUR: /* Handle RTCP FUR as FIR by setting the format to 4 */ rc = AST_RTP_RTCP_FMT_FIR; @@ -5981,7 +6132,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc ssrc = ntohl(rtpheader[2]); /* Determine the appropriate instance for this */ - child = rtp_find_instance_by_ssrc(instance, rtp, ssrc); + child = rtp_find_instance_by_packet_source_ssrc(instance, rtp, ssrc); if (!child) { /* Neither the bundled parent nor any child has this SSRC */ return &ast_null_frame; @@ -6614,6 +6765,8 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro } } else if (property == AST_RTP_PROPERTY_ASYMMETRIC_CODEC) { rtp->asymmetric_codec = value; + } else if (property == AST_RTP_PROPERTY_RETRANS_SEND) { + rtp->send_buffer = ast_data_buffer_alloc(ast_free_ptr, DEFAULT_RTP_BUFFER_SIZE); } } |