- OpenBSC - lists.osmocom.org

[PATCH 9/9] Fixed problem of mute audio on some calls
by Andreas Eversberg 16 Mar '12

16 Mar '12

When reading from RTP socket, the first read() may fail right after connecting to remote socket. Subsequent read() will work as it should. If the remote socket does not open fast enough, the transmitted RTP payload can cause an ICMP (connection refused) packet reply. This causes the read to fail with errno=111. In all other error cases, the errno is logged at debug level. In all error cases, reading is not disabled. --- openbsc/src/libtrau/rtp_proxy.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/openbsc/src/libtrau/rtp_proxy.c b/openbsc/src/libtrau/rtp_proxy.c index 88753b6..bb158e8 100644 --- a/openbsc/src/libtrau/rtp_proxy.c +++ b/openbsc/src/libtrau/rtp_proxy.c @@ -503,9 +503,16 @@ static int rtp_socket_read(struct rtp_socket *rs, struct rtp_sub_socket *rss) return -ENOMEM; rc = read(rss->bfd.fd, msg->data, RTP_ALLOC_SIZE); + if (rc == 0) + goto out_free; if (rc <= 0) { - rss->bfd.when &= ~BSC_FD_READ; - return rc; + /* Ignore "connection refused". this happens, If we open the + * socket faster than the remove side. */ + if (errno == 111) + return 0; + DEBUGPC(DLMUX, "Read of RTP socket (%p) failed (errno %d)\n", + rs, errno); + goto out_free; } msgb_put(msg, rc); -- 1.8.1.5 --------------050303020108080701030307--

1 0

Unknown payload type on OpenBSC when using rtp-bridge on LCR
by Nik Pakar 15 Mar '12

15 Mar '12

Hi Andreas, After some extensive testing i have done using jolly/rtpmux branch to get rtp to go directly between bsc and softswitch (yate in this case) i came across following scenario. MO call goes to OpenBSC/LCR then to yate routed back to OpenBSC/LCR and MT. Signalling seems all fine including codec negotiating as through out the path AMR/96 is been negotiated well. But once call connected, no rtp stream exist and i see following error on OpenBSC. received RTP frame with unknown payload type 98 Full topology and logs/trace from openbsc, LCR and yate is on this attachment. What could be the reason for bsc to get payload type 98. http://pastebin.com/rPfMamx0 Rgds Nik

3 2

Compiling LCR for OpenBSC
by Nik Pakar 15 Mar '12

15 Mar '12

Do I have to have mISDN for LCR even im not going to use any isdn interface ? Im trying to connect the NITB to LCR and LCR to asterisk all on ip. Thanks for any help. Rgds Nik

8 29

study on openbsc
by Kwan 13 Mar '12

13 Mar '12

Hello, I am a student from Hong Kong and I take a project about openbsc. I have a BTS that product from interWave. I contacted interWave engineer and he said he can borrow us the E1 card to connect BTS. But I don't know how to make openbsc to support this BTS. I can provide some detail about this BTS if you need. Thank, Kwan

2 3

nanoBTS behind NAT
by Nik Pakar 09 Mar '12

09 Mar '12

Dear all, will the openbsc setup works with nanoBTSs sitting behind a NAT ? Rgds Nik

5 10

[PATCH] Add support for AMR frames to MNCC/RTP interface
by Andreas Eversberg 08 Mar '12

08 Mar '12

AMR rate is currently fixed to 5.9k. --- openbsc/src/libmsc/gsm_04_08.c | 1 + openbsc/src/libtrau/rtp_proxy.c | 53 ++++++++++++++++++++++++++++++++++++----- 2 files changed, 48 insertions(+), 6 deletions(-) diff --git a/openbsc/src/libmsc/gsm_04_08.c b/openbsc/src/libmsc/gsm_04_08.c index df93433..a0650ab 100644 --- a/openbsc/src/libmsc/gsm_04_08.c +++ b/openbsc/src/libmsc/gsm_04_08.c @@ -2953,6 +2953,7 @@ int mncc_tx_to_cc(struct gsm_network *net, int msg_type, void *arg) case GSM_TCHF_FRAME: case GSM_TCHF_FRAME_EFR: case GSM_TCHH_FRAME: + case GSM_TCH_FRAME_AMR: /* Find callref */ trans = trans_find_by_callref(net, data->callref); if (!trans) { diff --git a/openbsc/src/libtrau/rtp_proxy.c b/openbsc/src/libtrau/rtp_proxy.c index 143bfa0..95729ef 100644 --- a/openbsc/src/libtrau/rtp_proxy.c +++ b/openbsc/src/libtrau/rtp_proxy.c @@ -192,21 +192,41 @@ static int rtp_decode(struct msgb *msg, uint32_t callref, struct msgb **data) return -EINVAL; } break; + case RTP_PT_AMR: + break; default: DEBUGPC(DLMUX, "received RTP frame with unknown payload " "type %d\n", rtph->payload_type); return -EINVAL; } - new_msg = msgb_alloc(sizeof(struct gsm_data_frame) + payload_len, - "GSM-DATA"); + if (payload_len > 33) { + DEBUGPC(DLMUX, "RTP payload too large (%d octets)\n", + payload_len); + return -EINVAL; + } + + if (rtph->payload_type == RTP_PT_AMR) { + new_msg = msgb_alloc(sizeof(struct gsm_data_frame) + 1 + + payload_len, "GSM-DATA (AMR)"); + } else { + new_msg = msgb_alloc(sizeof(struct gsm_data_frame) + + payload_len, "GSM-DATA"); + } if (!new_msg) return -ENOMEM; frame = (struct gsm_data_frame *)(new_msg->data); frame->msg_type = msg_type; frame->callref = callref; - memcpy(frame->data, payload, payload_len); - msgb_put(new_msg, sizeof(struct gsm_data_frame) + payload_len); + if (rtph->payload_type == RTP_PT_AMR) { + frame->data[0] = payload_len; + msgb_put(new_msg, sizeof(struct gsm_data_frame) + 1 + + payload_len); + memcpy(frame->data + 1, payload, payload_len); + } else { + msgb_put(new_msg, sizeof(struct gsm_data_frame) + payload_len); + memcpy(frame->data, payload, payload_len); + } *data = new_msg; return 0; @@ -264,6 +284,11 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) payload_len = RTP_LEN_GSM_HALF; duration = RTP_GSM_DURATION; break; + case GSM_TCH_FRAME_AMR: + payload_type = RTP_PT_AMR; + payload_len = frame->data[0]; + duration = RTP_GSM_DURATION; + break; default: DEBUGPC(DLMUX, "unsupported message type %d\n", frame->msg_type); @@ -291,7 +316,18 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) } } - msg = msgb_alloc(sizeof(struct rtp_hdr) + payload_len, "RTP-GSM-FULL"); + if (payload_len > 33) { + DEBUGPC(DLMUX, "RTP payload too large (%d octets)\n", + payload_len); + return -EINVAL; + } + + if (frame->msg_type == GSM_TCH_FRAME_AMR) + msg = msgb_alloc(sizeof(struct rtp_hdr) + payload_len, + "RTP-GSM (AMR)"); + else + msg = msgb_alloc(sizeof(struct rtp_hdr) + payload_len, + "RTP-GSM"); if (!msg) return -ENOMEM; rtph = (struct rtp_hdr *)msg->data; @@ -305,7 +341,12 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) rtph->timestamp = htonl(rs->transmit.timestamp); rs->transmit.timestamp += duration; rtph->ssrc = htonl(rs->transmit.ssrc); - memcpy(msg->data + sizeof(struct rtp_hdr), frame->data, payload_len); + if (frame->msg_type == GSM_TCH_FRAME_AMR) + memcpy(msg->data + sizeof(struct rtp_hdr), frame->data + 1, + payload_len); + else + memcpy(msg->data + sizeof(struct rtp_hdr), frame->data, + payload_len); msgb_put(msg, sizeof(struct rtp_hdr) + payload_len); msgb_enqueue(&rss->tx_queue, msg); rss->bfd.when |= BSC_FD_WRITE; -- 1.8.1.5 --------------040208000603050009060705--

1 0

[PATCH] Add support for AMR frames to MNCC/RTP interface
by Andreas Eversberg 08 Mar '12

08 Mar '12

AMR rate is currently fixed to 5.9k. --- openbsc/src/libmsc/gsm_04_08.c | 1 + openbsc/src/libtrau/rtp_proxy.c | 35 ++++++++++++++++++++++++++++++----- 2 files changed, 31 insertions(+), 5 deletions(-) diff --git a/openbsc/src/libmsc/gsm_04_08.c b/openbsc/src/libmsc/gsm_04_08.c index df93433..a0650ab 100644 --- a/openbsc/src/libmsc/gsm_04_08.c +++ b/openbsc/src/libmsc/gsm_04_08.c @@ -2953,6 +2953,7 @@ int mncc_tx_to_cc(struct gsm_network *net, int msg_type, void *arg) case GSM_TCHF_FRAME: case GSM_TCHF_FRAME_EFR: case GSM_TCHH_FRAME: + case GSM_TCH_FRAME_AMR: /* Find callref */ trans = trans_find_by_callref(net, data->callref); if (!trans) { diff --git a/openbsc/src/libtrau/rtp_proxy.c b/openbsc/src/libtrau/rtp_proxy.c index 143bfa0..913d64d 100644 --- a/openbsc/src/libtrau/rtp_proxy.c +++ b/openbsc/src/libtrau/rtp_proxy.c @@ -192,21 +192,35 @@ static int rtp_decode(struct msgb *msg, uint32_t callref, struct msgb **data) return -EINVAL; } break; + case RTP_PT_AMR: + break; default: DEBUGPC(DLMUX, "received RTP frame with unknown payload " "type %d\n", rtph->payload_type); return -EINVAL; } - new_msg = msgb_alloc(sizeof(struct gsm_data_frame) + payload_len, - "GSM-DATA"); + if (rtph->payload_type == RTP_PT_AMR) { + new_msg = msgb_alloc(sizeof(struct gsm_data_frame) + 1 + + payload_len, "GSM-DATA"); + } else { + new_msg = msgb_alloc(sizeof(struct gsm_data_frame) + + payload_len, "GSM-DATA"); + } if (!new_msg) return -ENOMEM; frame = (struct gsm_data_frame *)(new_msg->data); frame->msg_type = msg_type; frame->callref = callref; - memcpy(frame->data, payload, payload_len); - msgb_put(new_msg, sizeof(struct gsm_data_frame) + payload_len); + if (rtph->payload_type == RTP_PT_AMR) { + frame->data[0] = payload_len; + msgb_put(new_msg, sizeof(struct gsm_data_frame) + 1 + + payload_len); + memcpy(frame->data + 1, payload, payload_len); + } else { + msgb_put(new_msg, sizeof(struct gsm_data_frame) + payload_len); + memcpy(frame->data, payload, payload_len); + } *data = new_msg; return 0; @@ -264,6 +278,11 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) payload_len = RTP_LEN_GSM_HALF; duration = RTP_GSM_DURATION; break; + case GSM_TCH_FRAME_AMR: + payload_type = RTP_PT_AMR; + payload_len = frame->data[0]; + duration = RTP_GSM_DURATION; + break; default: DEBUGPC(DLMUX, "unsupported message type %d\n", frame->msg_type); @@ -305,7 +324,13 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) rtph->timestamp = htonl(rs->transmit.timestamp); rs->transmit.timestamp += duration; rtph->ssrc = htonl(rs->transmit.ssrc); - memcpy(msg->data + sizeof(struct rtp_hdr), frame->data, payload_len); + if (frame->msg_type == GSM_TCH_FRAME_AMR) { + memcpy(msg->data + sizeof(struct rtp_hdr), frame->data + 1, + payload_len); + } else { + memcpy(msg->data + sizeof(struct rtp_hdr), frame->data, + payload_len); + } msgb_put(msg, sizeof(struct rtp_hdr) + payload_len); msgb_enqueue(&rss->tx_queue, msg); rss->bfd.when |= BSC_FD_WRITE; -- 1.8.3.2 --------------090303090405010206070901--

1 0

[PATCH 8/9] Fixed (ipaccess BTS) delay/silence problems, if RTP stream jitters too much
by Andreas Eversberg 08 Mar '12

08 Mar '12

After OpenBSC stalled for some reason (e.g. CPU overload or database access) or after speech frames have been lost (MNCC application problems / hold/retrieve call), the timestamp and the sequence number of the RTP socket state must be corrected. The amount of incrmentation is calculated from the elapsed time. Not incrementing timestamp and sequence number would cause all frames to be dropped by ipaccess BTS, because the BTS expects frames with more recent timestamps. If speech frames are received too fast, they must be dropped. The timestamp and sequence number of the RTP socket state are not changed in this case. Incmenetating timestamp and sequence number would causes high delay at ipaccess BTS, because the BTS would queue the frames until the timestamp matches the current time. There is a simple test case: Make a call between two phones and check the delay. (When using LCR, make a call to the echo test.) The press CTRL+z to suspend OpenBSC process for a few seconds and enter "fg" to continue OpenBSC process. There shall be no change in the delay, even after repeating this test many times. --- openbsc/src/libtrau/rtp_proxy.c | 89 +++++++++++++++++++++++++++-------------- 1 file changed, 60 insertions(+), 29 deletions(-) diff --git a/openbsc/src/libtrau/rtp_proxy.c b/openbsc/src/libtrau/rtp_proxy.c index 0c4d82b..fa6b629 100644 --- a/openbsc/src/libtrau/rtp_proxy.c +++ b/openbsc/src/libtrau/rtp_proxy.c @@ -236,19 +236,65 @@ static int rtp_decode(struct msgb *msg, uint32_t callref, struct msgb **data, in return 0; } -/* "to - from" */ -static void tv_difference(struct timeval *diff, const struct timeval *from, - const struct timeval *__to) +#define USEC_1S 1000000 +#define USEC_10MS 10000 +#define USEC_20MS 20000 +#define SAMPLES_1S 8000 +#define USEC_SAMPLE 125 + +/* add usec to tv */ +static void tv_add_usec(struct timeval *tv, long int usec) { - struct timeval _to = *__to, *to = &_to; + struct timeval tv_add; - if (to->tv_usec < from->tv_usec) { - to->tv_sec -= 1; - to->tv_usec += 1000000; + tv_add.tv_sec = usec / USEC_1S; + tv_add.tv_usec = usec % USEC_1S; + timeradd(tv, &tv_add, tv); +} + +static int correct_timestamp(struct rtp_socket *rs, int duration) +{ + struct timeval tv, tv_diff; + long int usec_diff, frame_diff; + int usec_duration = duration * USEC_SAMPLE; + + gettimeofday(&tv, NULL); + timersub(&tv, &rs->transmit.last_tv, &tv_diff); + + usec_diff = tv_diff.tv_sec * USEC_1S + tv_diff.tv_usec; + frame_diff = (usec_diff + (usec_duration >> 1)) / usec_duration; /* round */ + + /* Drop frame, if current time to too much in advance of the last_tv. + * < 0 means that the time difference in frames must be at lease 2 + * frames below the expected difference of 1. + */ + if (frame_diff < 0) + return -1; + + /* Increment last_tv by the duration of one frame. */ + tv_add_usec(&rs->transmit.last_tv, usec_duration); + + /* Increment last_tv, if the current time is too much afterwards. + * Also increment timestamp and sequence number of RTP socket state. + * > 2 means that the time difference in frames must be at least 2 + * frames above the expected difference of 1. + */ + if (frame_diff > 2) { + long int frame_diff_excess = frame_diff - 1; + long int sample_diff_excess = frame_diff_excess * duration; + + /* correct last_tv */ + tv_add_usec(&rs->transmit.last_tv, + sample_diff_excess * USEC_SAMPLE); + LOGP(DLMUX, LOGL_NOTICE, + "Correcting timestamp difference of %ld frames " + "(to %s)\n", frame_diff_excess, + (rs->rx_action == RTP_RECV_APP) ? "app" : "BTS"); + rs->transmit.sequence += frame_diff_excess; + rs->transmit.timestamp += sample_diff_excess; } - diff->tv_usec = to->tv_usec - from->tv_usec; - diff->tv_sec = to->tv_sec - from->tv_sec; + return 0; } /*! \brief encode and send a rtp frame @@ -265,6 +311,7 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) int duration; /* in samples */ int is_amr = 0, is_bfi = 0; uint8_t dynamic_pt = 0; + int rc; if (rs->rx_action == RTP_RECV_APP) dynamic_pt = rs->receive.payload_type; @@ -275,6 +322,7 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) rs->transmit.ssrc = rand(); rs->transmit.sequence = random(); rs->transmit.timestamp = random(); + gettimeofday(&rs->transmit.last_tv, NULL); } switch (frame->msg_type) { @@ -311,26 +359,9 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) return -EINVAL; } - { - struct timeval tv, tv_diff; - long int usec_diff, frame_diff; - - gettimeofday(&tv, NULL); - tv_difference(&tv_diff, &rs->transmit.last_tv, &tv); - rs->transmit.last_tv = tv; - - usec_diff = tv_diff.tv_sec * 1000000 + tv_diff.tv_usec; - frame_diff = (usec_diff / 20000); - - if (abs(frame_diff) > 1) { - long int frame_diff_excess = frame_diff - 1; - - LOGP(DLMUX, LOGL_NOTICE, - "Correcting frame difference of %ld frames\n", frame_diff_excess); - rs->transmit.sequence += frame_diff_excess; - rs->transmit.timestamp += frame_diff_excess * duration; - } - } + rc = correct_timestamp(rs, duration); + if (rc) + return 0; if (is_bfi) { /* In case of a bad frame, just count and drop packt. */ -- 1.8.1.5 --------------070804050601000205010809--

1 0

[PATCH 8/9] Fixed (nanoBTS) delay problems, if RTP stream jitters too much
by Andreas Eversberg 08 Mar '12

08 Mar '12

The RTP stream is generated or forwarded by OpenBSC to nanoBTS. Due to switching of streams (hold/retrieve call), packet loss or even stalling of sender's process, the time stamp must be corrected. If outdated packets are received, they get dropped. --- openbsc/src/libtrau/rtp_proxy.c | 82 ++++++++++++++++++++++++++++++----------- 1 file changed, 61 insertions(+), 21 deletions(-) diff --git a/openbsc/src/libtrau/rtp_proxy.c b/openbsc/src/libtrau/rtp_proxy.c index 2284798..88753b6 100644 --- a/openbsc/src/libtrau/rtp_proxy.c +++ b/openbsc/src/libtrau/rtp_proxy.c @@ -236,6 +236,12 @@ static int rtp_decode(struct msgb *msg, uint32_t callref, struct msgb **data, in return 0; } +#define USEC_1S 1000000 +#define USEC_10MS 10000 +#define USEC_20MS 20000 +#define SAMPLES_1S 8000 +#define USEC_SAMPLE 125 + /* "to - from" */ static void tv_difference(struct timeval *diff, const struct timeval *from, const struct timeval *__to) @@ -244,13 +250,62 @@ static void tv_difference(struct timeval *diff, const struct timeval *from, if (to->tv_usec < from->tv_usec) { to->tv_sec -= 1; - to->tv_usec += 1000000; + to->tv_usec += USEC_1S; } diff->tv_usec = to->tv_usec - from->tv_usec; diff->tv_sec = to->tv_sec - from->tv_sec; } +/* add sec,usec to tv */ +static void tv_add(struct timeval *tv, int sec, int usec) +{ + + while (usec < 0) { + usec += USEC_1S; + sec--; + } + tv->tv_sec += sec; + tv->tv_usec += usec; + while (tv->tv_usec >= USEC_1S) { + tv->tv_sec++; + tv->tv_usec -= USEC_1S; + } +} + +static int correct_timestamp(struct rtp_socket *rs, int duration) +{ + struct timeval tv, tv_diff; + long int usec_diff, frame_diff; + + gettimeofday(&tv, NULL); + tv_difference(&tv_diff, &rs->transmit.last_tv, &tv); + tv_add(&rs->transmit.last_tv, 0, USEC_20MS); + + usec_diff = tv_diff.tv_sec * USEC_1S + tv_diff.tv_usec; + frame_diff = ((usec_diff + (USEC_10MS)) / USEC_20MS); /* round */ + + if (abs(frame_diff - 1) > 1) { + long int frame_diff_excess = frame_diff - 1; + long int sample_diff_excess = frame_diff_excess * duration; + + /* correct last_tv */ + tv_add(&rs->transmit.last_tv, sample_diff_excess / SAMPLES_1S, + (sample_diff_excess % SAMPLES_1S) * USEC_SAMPLE); + /* drop frame, if time stamp is in the past */ + if (frame_diff_excess < 0) + return -1; + LOGP(DLMUX, LOGL_NOTICE, + "Correcting timestamp difference of %ld frames " + "(to %s)\n", frame_diff_excess, + (rs->rx_action == RTP_RECV_L4) ? "app" : "BTS"); + rs->transmit.sequence += frame_diff_excess; + rs->transmit.timestamp += sample_diff_excess; + } + + return 0; +} + /*! \brief encode and send a rtp frame * \param[in] rs RTP socket through which we shall send * \param[in] frame GSM RTP frame to be sent @@ -265,6 +320,7 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) int duration; /* in samples */ int amr = 0; uint8_t dynamic_pt = 0; + int rc; if (rs->rx_action == RTP_RECV_L4) dynamic_pt = rs->receive.payload_type; @@ -275,6 +331,7 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) rs->transmit.ssrc = rand(); rs->transmit.sequence = random(); rs->transmit.timestamp = random(); + gettimeofday(&rs->transmit.last_tv, NULL); } switch (frame->msg_type) { @@ -313,26 +370,9 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) return -EINVAL; } - { - struct timeval tv, tv_diff; - long int usec_diff, frame_diff; - - gettimeofday(&tv, NULL); - tv_difference(&tv_diff, &rs->transmit.last_tv, &tv); - rs->transmit.last_tv = tv; - - usec_diff = tv_diff.tv_sec * 1000000 + tv_diff.tv_usec; - frame_diff = (usec_diff / 20000); - - if (abs(frame_diff) > 1) { - long int frame_diff_excess = frame_diff - 1; - - LOGP(DLMUX, LOGL_NOTICE, - "Correcting frame difference of %ld frames\n", frame_diff_excess); - rs->transmit.sequence += frame_diff_excess; - rs->transmit.timestamp += frame_diff_excess * duration; - } - } + rc = correct_timestamp(rs, duration); + if (rc) + return 0; if (frame->msg_type == GSM_BAD_FRAME) return 0; -- 1.8.1.5 --------------000504080100090805040102--

1 0

[PATCH 8/9] Fixed (nanoBTS) delay problems, if RTP stream jitters too much
by Andreas Eversberg 08 Mar '12

08 Mar '12

The RTP stream is generated or forwarded by OpenBSC to nanoBTS. Due to switching of streams (hold/retrieve call), packet loss or even stalling of sender's process, the time stamp must be corrected. If outdated packets are received, they get dropped. --- openbsc/src/libtrau/rtp_proxy.c | 80 ++++++++++++++++++++++++++++++----------- 1 file changed, 59 insertions(+), 21 deletions(-) diff --git a/openbsc/src/libtrau/rtp_proxy.c b/openbsc/src/libtrau/rtp_proxy.c index ed7479d..3e6c462 100644 --- a/openbsc/src/libtrau/rtp_proxy.c +++ b/openbsc/src/libtrau/rtp_proxy.c @@ -236,6 +236,12 @@ static int rtp_decode(struct msgb *msg, uint32_t callref, struct msgb **data, in return 0; } +#define USEC_1S 1000000 +#define USEC_10MS 10000 +#define USEC_20MS 20000 +#define SAMPLES_1S 8000 +#define USEC_SAMPLE 125 + /* "to - from" */ static void tv_difference(struct timeval *diff, const struct timeval *from, const struct timeval *__to) @@ -244,13 +250,60 @@ static void tv_difference(struct timeval *diff, const struct timeval *from, if (to->tv_usec < from->tv_usec) { to->tv_sec -= 1; - to->tv_usec += 1000000; + to->tv_usec += USEC_1S; } diff->tv_usec = to->tv_usec - from->tv_usec; diff->tv_sec = to->tv_sec - from->tv_sec; } +/* add sec,usec to tv */ +static void tv_add(struct timeval *tv, int sec, int usec) +{ + + if (usec < 0) + usec += USEC_1S; + tv->tv_sec += sec; + tv->tv_usec += usec; + if (tv->tv_usec >= USEC_1S) { + tv->tv_sec++; + tv->tv_usec -= USEC_1S; + } +} + +static int correct_timestamp(struct rtp_socket *rs, int duration) +{ + struct timeval tv, tv_diff; + long int usec_diff, frame_diff; + + gettimeofday(&tv, NULL); + tv_difference(&tv_diff, &rs->transmit.last_tv, &tv); + tv_add(&rs->transmit.last_tv, 0, USEC_20MS); + + usec_diff = tv_diff.tv_sec * USEC_1S + tv_diff.tv_usec; + frame_diff = ((usec_diff + (USEC_10MS)) / USEC_20MS); /* round */ + + if (abs(frame_diff - 1) > 1) { + long int frame_diff_excess = frame_diff - 1; + long int sample_diff_excess = frame_diff_excess * duration; + + /* correct last_tv */ + tv_add(&rs->transmit.last_tv, sample_diff_excess / SAMPLES_1S, + (sample_diff_excess % SAMPLES_1S) * USEC_SAMPLE); + /* drop frame, if time stamp is in the past */ + if (frame_diff_excess < 0) + return -1; + LOGP(DLMUX, LOGL_NOTICE, + "Correcting timestamp difference of %ld frames " + "(to %s)\n", frame_diff_excess, + (rs->rx_action == RTP_RECV_L4) ? "app" : "BTS"); + rs->transmit.sequence += frame_diff_excess; + rs->transmit.timestamp += sample_diff_excess; + } + + return 0; +} + /*! \brief encode and send a rtp frame * \param[in] rs RTP socket through which we shall send * \param[in] frame GSM RTP frame to be sent @@ -265,6 +318,7 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) int duration; /* in samples */ int amr = 0; uint8_t dynamic_pt = 0; + int rc; if (rs->rx_action == RTP_RECV_L4) dynamic_pt = rs->receive.payload_type; @@ -275,6 +329,7 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) rs->transmit.ssrc = rand(); rs->transmit.sequence = random(); rs->transmit.timestamp = random(); + gettimeofday(&rs->transmit.last_tv, NULL); } switch (frame->msg_type) { @@ -313,26 +368,9 @@ int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) return -EINVAL; } - { - struct timeval tv, tv_diff; - long int usec_diff, frame_diff; - - gettimeofday(&tv, NULL); - tv_difference(&tv_diff, &rs->transmit.last_tv, &tv); - rs->transmit.last_tv = tv; - - usec_diff = tv_diff.tv_sec * 1000000 + tv_diff.tv_usec; - frame_diff = (usec_diff / 20000); - - if (abs(frame_diff) > 1) { - long int frame_diff_excess = frame_diff - 1; - - LOGP(DLMUX, LOGL_NOTICE, - "Correcting frame difference of %ld frames\n", frame_diff_excess); - rs->transmit.sequence += frame_diff_excess; - rs->transmit.timestamp += frame_diff_excess * duration; - } - } + rc = correct_timestamp(rs, duration); + if (rc) + return 0; if (frame->msg_type == GSM_BAD_FRAME) return 0; -- 1.8.1.5 --------------090106010400060202050508--

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