/* $Id$ */ /* * Copyright (C) 2003-2007 Benny Prijono * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include #include #include #include static const char *USAGE = "pcaputil [options] INPUT OUTPUT\n" "\n" " Convert captured RTP packets in PCAP file to WAV or stream it\n" " to remote destination.\n" "\n" "INPUT is the PCAP file name/path\n" "\n" "Options to filter packets from PCAP file:\n" "(you can always select the relevant packets from Wireshark of course!)\n" " --src-ip=IP Only include packets from this source address\n" " --dst-ip=IP Only include packets destined to this address\n" " --src-port=port Only include packets from this source port number\n" " --dst-port=port Only include packets destined to this port number\n" "\n" "Options for saving to WAV file:\n" "" " OUTPUT is WAV file: Set output to WAV file. The program will decode the\n" " RTP contents to the specified WAV file using codec\n" " that is available in PJMEDIA, and optionally decrypt\n" " the content using the SRTP crypto and keys below.\n" " --srtp-crypto=TAG, -c Set crypto to be used to decrypt SRTP packets. Valid\n" " tags are: \n" " AES_CM_128_HMAC_SHA1_80 \n" " AES_CM_128_HMAC_SHA1_32\n" " --srtp-key=KEY, -k Set the base64 key to decrypt SRTP packets.\n" "\n" " Example:\n" " pcaputil file.pcap output.wav\n" " pcaputil -c AES_CM_128_HMAC_SHA1_80 \\\n" " -k VLDONbsbGl2Puqy+0PV7w/uGfpSPKFevDpxGsxN3 \\\n" " file.pcap output.wav\n" "\n" "Remote streaming is not supported yet." ; static struct app { pj_caching_pool cp; pj_pool_t *pool; pjmedia_endpt *mept; pj_pcap_file *pcap; pjmedia_port *wav; pjmedia_codec *codec; unsigned pt; pjmedia_transport *srtp; pjmedia_rtp_session rtp_sess; pj_bool_t rtp_sess_init; } app; static void err_exit(const char *title, pj_status_t status) { if (status != PJ_SUCCESS) { char errmsg[PJ_ERR_MSG_SIZE]; pj_strerror(status, errmsg, sizeof(errmsg)); printf("Error: %s: %s\n", title, errmsg); } else { printf("Error: %s\n", title); } if (app.srtp) pjmedia_transport_close(app.srtp); if (app.wav) { pj_ssize_t pos = pjmedia_wav_writer_port_get_pos(app.wav); if (pos >= 0) { unsigned msec; msec = pos / 2 * 1000 / app.wav->info.clock_rate; printf("Written: %dm:%02ds.%03d\n", msec / 1000 / 60, (msec / 1000) % 60, msec % 1000); } pjmedia_port_destroy(app.wav); } if (app.pcap) pj_pcap_close(app.pcap); if (app.codec) { pjmedia_codec_mgr *cmgr; app.codec->op->close(app.codec); cmgr = pjmedia_endpt_get_codec_mgr(app.mept); pjmedia_codec_mgr_dealloc_codec(cmgr, app.codec); } if (app.mept) pjmedia_endpt_destroy(app.mept); if (app.pool) pj_pool_release(app.pool); pj_caching_pool_destroy(&app.cp); pj_shutdown(); exit(1); } #define T(op) do { \ status = op; \ if (status != PJ_SUCCESS) \ err_exit(#op, status); \ } while (0) static void read_rtp(pj_uint8_t *buf, pj_size_t bufsize, pjmedia_rtp_hdr **rtp, pj_uint8_t **payload, unsigned *payload_size) { pj_status_t status; /* Init RTP session */ if (!app.rtp_sess_init) { T(pjmedia_rtp_session_init(&app.rtp_sess, 0, 0)); app.rtp_sess_init = PJ_TRUE; } /* Loop reading until we have a good RTP packet */ for (;;) { pj_size_t sz = bufsize; const pjmedia_rtp_hdr *r; const void *p; pjmedia_rtp_status seq_st; status = pj_pcap_read_udp(app.pcap, NULL, buf, &sz); if (status != PJ_SUCCESS) err_exit("Error reading PCAP file", status); /* Decode RTP packet to make sure that this is an RTP packet. * We will decode it again to get the payload after we do * SRTP decoding */ status = pjmedia_rtp_decode_rtp(&app.rtp_sess, buf, sz, &r, &p, payload_size); if (status != PJ_SUCCESS) { char errmsg[PJ_ERR_MSG_SIZE]; pj_strerror(status, errmsg, sizeof(errmsg)); printf("Not RTP packet, skipping packet: %s\n", errmsg); continue; } /* Decrypt SRTP */ #if PJMEDIA_HAS_SRTP if (app.srtp) { int len = sz; status = pjmedia_transport_srtp_decrypt_pkt(app.srtp, PJ_TRUE, buf, &len); if (status != PJ_SUCCESS) { char errmsg[PJ_ERR_MSG_SIZE]; pj_strerror(status, errmsg, sizeof(errmsg)); printf("SRTP packet decryption failed, skipping packet: %s\n", errmsg); continue; } sz = len; /* Decode RTP packet again */ status = pjmedia_rtp_decode_rtp(&app.rtp_sess, buf, sz, &r, &p, payload_size); if (status != PJ_SUCCESS) { char errmsg[PJ_ERR_MSG_SIZE]; pj_strerror(status, errmsg, sizeof(errmsg)); printf("Not RTP packet, skipping packet: %s\n", errmsg); continue; } } #endif /* Update RTP session */ pjmedia_rtp_session_update(&app.rtp_sess, r, &seq_st); /* Skip out-of-order packet */ if (seq_st.diff == 0) { printf("Skipping out of order packet\n"); continue; } /* Skip if payload type is different */ if (r->pt != app.pt) { printf("Skipping RTP packet with bad payload type\n"); continue; } /* Skip bad packet */ if (seq_st.status.flag.bad) { printf("Skipping bad RTP\n"); continue; } *rtp = (pjmedia_rtp_hdr*)r; *payload = (pj_uint8_t*)p; /* We have good packet */ break; } } static void pcap2wav(const char *wav_filename, const pj_str_t *srtp_crypto, const pj_str_t *srtp_key) { struct pkt { pj_uint8_t buffer[320]; pjmedia_rtp_hdr *rtp; pj_uint8_t *payload; unsigned payload_len; } pkt0; pjmedia_codec_mgr *cmgr; const pjmedia_codec_info *ci; pjmedia_codec_param param; unsigned samples_per_frame; pj_status_t status; /* Initialize all codecs */ #if PJMEDIA_HAS_SPEEX_CODEC T( pjmedia_codec_speex_init(app.mept, 0, 10, 10) ); #endif /* PJMEDIA_HAS_SPEEX_CODEC */ #if PJMEDIA_HAS_ILBC_CODEC T( pjmedia_codec_ilbc_init(app.mept, 30) ); #endif /* PJMEDIA_HAS_ILBC_CODEC */ #if PJMEDIA_HAS_GSM_CODEC T( pjmedia_codec_gsm_init(app.mept) ); #endif /* PJMEDIA_HAS_GSM_CODEC */ #if PJMEDIA_HAS_G711_CODEC T( pjmedia_codec_g711_init(app.mept) ); #endif /* PJMEDIA_HAS_G711_CODEC */ #if PJMEDIA_HAS_G722_CODEC T( pjmedia_codec_g722_init(app.mept) ); #endif /* PJMEDIA_HAS_G722_CODEC */ #if PJMEDIA_HAS_L16_CODEC T( pjmedia_codec_l16_init(app.mept, 0) ); #endif /* PJMEDIA_HAS_L16_CODEC */ /* Create SRTP transport is needed */ #if PJMEDIA_HAS_SRTP if (srtp_crypto->slen) { pjmedia_srtp_crypto crypto; pj_bzero(&crypto, sizeof(crypto)); crypto.key = *srtp_key; crypto.name = *srtp_crypto; T( pjmedia_transport_srtp_create(app.mept, NULL, NULL, &app.srtp) ); T( pjmedia_transport_srtp_start(app.srtp, &crypto, &crypto) ); } #else PJ_UNUSED_ARG(srtp_crypto); PJ_UNUSED_ARG(srtp_key); #endif /* Read first packet */ read_rtp(pkt0.buffer, sizeof(pkt0.buffer), &pkt0.rtp, &pkt0.payload, &pkt0.payload_len); cmgr = pjmedia_endpt_get_codec_mgr(app.mept); /* Get codec info and param for the specified payload type */ app.pt = pkt0.rtp->pt; T( pjmedia_codec_mgr_get_codec_info(cmgr, pkt0.rtp->pt, &ci) ); T( pjmedia_codec_mgr_get_default_param(cmgr, ci, ¶m) ); /* Alloc and init codec */ T( pjmedia_codec_mgr_alloc_codec(cmgr, ci, &app.codec) ); T( app.codec->op->init(app.codec, app.pool) ); T( app.codec->op->open(app.codec, ¶m) ); /* Open WAV file */ samples_per_frame = ci->clock_rate * param.info.frm_ptime / 1000; T( pjmedia_wav_writer_port_create(app.pool, wav_filename, ci->clock_rate, ci->channel_cnt, samples_per_frame, param.info.pcm_bits_per_sample, 0, 0, &app.wav) ); /* Loop reading PCAP and writing WAV file */ for (;;) { struct pkt pkt1; pj_timestamp ts; pjmedia_frame frames[16], pcm_frame; short pcm[320]; unsigned i, frame_cnt; long samples_cnt, ts_gap; pj_assert(sizeof(pcm) >= samples_per_frame); /* Parse first packet */ ts.u64 = 0; frame_cnt = PJ_ARRAY_SIZE(frames); T( app.codec->op->parse(app.codec, pkt0.payload, pkt0.payload_len, &ts, &frame_cnt, frames) ); /* Decode and write to WAV file */ samples_cnt = 0; for (i=0; iop->decode(app.codec, &frames[i], pcm_frame.size, &pcm_frame) ); T( pjmedia_port_put_frame(app.wav, &pcm_frame) ); samples_cnt += samples_per_frame; } /* Read next packet */ read_rtp(pkt1.buffer, sizeof(pkt1.buffer), &pkt1.rtp, &pkt1.payload, &pkt1.payload_len); /* Fill in the gap (if any) between pkt0 and pkt1 */ ts_gap = pj_ntohl(pkt1.rtp->ts) - pj_ntohl(pkt0.rtp->ts) - samples_cnt; while (ts_gap >= (long)samples_per_frame) { pcm_frame.buf = pcm; pcm_frame.size = samples_per_frame * 2; if (app.codec->op->recover) { T( app.codec->op->recover(app.codec, pcm_frame.size, &pcm_frame) ); } else { pj_bzero(pcm_frame.buf, pcm_frame.size); } T( pjmedia_port_put_frame(app.wav, &pcm_frame) ); ts_gap -= samples_per_frame; } /* Next */ pkt0 = pkt1; pkt0.rtp = (pjmedia_rtp_hdr*)pkt0.buffer; pkt0.payload = pkt0.buffer + (pkt1.payload - pkt1.buffer); } } int main(int argc, char *argv[]) { pj_str_t input, output, wav, srtp_crypto, srtp_key; pj_pcap_filter filter; pj_status_t status; enum { OPT_SRC_IP = 1, OPT_DST_IP, OPT_SRC_PORT, OPT_DST_PORT }; struct pj_getopt_option long_options[] = { { "srtp-crypto", 1, 0, 'c' }, { "srtp-key", 1, 0, 'k' }, { "src-ip", 1, 0, OPT_SRC_IP }, { "dst-ip", 1, 0, OPT_DST_IP }, { "src-port", 1, 0, OPT_SRC_PORT }, { "dst-port", 1, 0, OPT_DST_PORT }, { NULL, 0, 0, 0} }; int c; int option_index; char key_bin[32]; srtp_crypto.slen = srtp_key.slen = 0; pj_pcap_filter_default(&filter); filter.link = PJ_PCAP_LINK_TYPE_ETH; filter.proto = PJ_PCAP_PROTO_TYPE_UDP; /* Parse arguments */ pj_optind = 0; while((c=pj_getopt_long(argc,argv, "c:k:", long_options, &option_index))!=-1) { switch (c) { case 'c': srtp_crypto = pj_str(pj_optarg); break; case 'k': { int key_len = sizeof(key_bin); srtp_key = pj_str(pj_optarg); if (pj_base64_decode(&srtp_key, (pj_uint8_t*)key_bin, &key_len)) { puts("Error: invalid key"); return 1; } srtp_key.ptr = key_bin; srtp_key.slen = key_len; } break; case OPT_SRC_IP: { pj_str_t t = pj_str(pj_optarg); pj_in_addr a = pj_inet_addr(&t); filter.ip_src = a.s_addr; } break; case OPT_DST_IP: { pj_str_t t = pj_str(pj_optarg); pj_in_addr a = pj_inet_addr(&t); filter.ip_dst = a.s_addr; } break; case OPT_SRC_PORT: filter.src_port = pj_htons((pj_uint16_t)atoi(pj_optarg)); break; case OPT_DST_PORT: filter.dst_port = pj_htons((pj_uint16_t)atoi(pj_optarg)); break; default: puts("Error: invalid option"); return 1; } } if (pj_optind != argc - 2) { puts(USAGE); return 1; } if (!(srtp_crypto.slen) != !(srtp_key.slen)) { puts("Error: both SRTP crypto and key must be specified"); puts(USAGE); return 1; } input = pj_str(argv[pj_optind]); output = pj_str(argv[pj_optind+1]); wav = pj_str(".wav"); T( pj_init() ); pj_caching_pool_init(&app.cp, NULL, 0); app.pool = pj_pool_create(&app.cp.factory, "pcaputil", 1000, 1000, NULL); T( pjlib_util_init() ); T( pjmedia_endpt_create(&app.cp.factory, NULL, 0, &app.mept) ); T( pj_pcap_open(app.pool, input.ptr, &app.pcap) ); T( pj_pcap_set_filter(app.pcap, &filter) ); if (pj_stristr(&output, &wav)) { pcap2wav(output.ptr, &srtp_crypto, &srtp_key); } else { err_exit("invalid output file", PJ_EINVAL); } pjmedia_endpt_destroy(app.mept); pj_pool_release(app.pool); pj_caching_pool_destroy(&app.cp); pj_shutdown(); return 0; }