diff --git a/jni/pjproject-android/pjsip-apps/src/samples/pcaputil.c b/jni/pjproject-android/pjsip-apps/src/samples/pcaputil.c
new file mode 100644
index 0000000..3eb1603
--- /dev/null
+++ b/jni/pjproject-android/pjsip-apps/src/samples/pcaputil.c
@@ -0,0 +1,540 @@
+/* $Id: pcaputil.c 4537 2013-06-19 06:47:43Z riza $ */
+/* 
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * 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 <pjlib.h>
+#include <pjlib-util.h>
+#include <pjmedia.h>
+#include <pjmedia-codec.h>
+
+static const char *USAGE =
+"pcaputil [options] INPUT OUTPUT\n"
+"\n"
+"  Convert captured RTP packets in PCAP file to WAV file or play it\n"
+"  to audio device.\n"
+"\n"
+"  INPUT  is the PCAP file name/path.\n"
+"  OUTPUT is the WAV file name/path to store the output, or set to \"-\",\n"
+"         to play the output to audio device. The program will decode\n"
+"         the RTP contents using codec that is available in PJMEDIA,\n"
+"         and optionally decrypt the content using the SRTP crypto and\n"
+"         keys below.\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 RTP packet processing:\n"
+""
+"  --codec=codec_id	  The codec ID formatted \"name/clock-rate/channel-count\"\n"
+"                         must be specified for codec with dynamic PT,\n"
+"                         e.g: \"Speex/8000\"\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"
+"Options for playing to audio device:\n"
+""
+"  --play-dev-id=dev_id   Audio device ID for playback.\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"
+;
+
+static struct app
+{
+    pj_caching_pool	 cp;
+    pj_pool_t		*pool;
+    pjmedia_endpt	*mept;
+    pj_pcap_file	*pcap;
+    pjmedia_port	*wav;
+    pjmedia_codec	*codec;
+    pjmedia_aud_stream  *aud_strm;
+    unsigned		 pt;
+    pjmedia_transport	*srtp;
+    pjmedia_rtp_session	 rtp_sess;
+    pj_bool_t		 rtp_sess_init;
+} app;
+
+
+static void cleanup()
+{
+    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 = (unsigned)pos / 2 * 1000 / PJMEDIA_PIA_SRATE(&app.wav->info);
+            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;
+	pjmedia_codec_close(app.codec);
+	cmgr = pjmedia_endpt_get_codec_mgr(app.mept);
+	pjmedia_codec_mgr_dealloc_codec(cmgr, app.codec);
+    }
+    if (app.aud_strm) {
+	pjmedia_aud_stream_stop(app.aud_strm);
+	pjmedia_aud_stream_destroy(app.aud_strm);
+    }
+    if (app.mept) pjmedia_endpt_destroy(app.mept);
+    if (app.pool) pj_pool_release(app.pool);
+    pj_caching_pool_destroy(&app.cp);
+    pj_shutdown();
+}
+
+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);
+    }
+    cleanup();
+    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_bool_t check_pt)
+{
+    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, (int)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 = (int)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, (int)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_update2(&app.rtp_sess, r, &seq_st, PJ_FALSE);
+
+	/* 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 (check_pt && 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;
+    }
+}
+
+pjmedia_frame play_frm;
+static pj_bool_t play_frm_copied, play_frm_ready;
+
+static pj_status_t wait_play(pjmedia_frame *f)
+{
+    play_frm_copied = PJ_FALSE;
+    play_frm = *f;
+    play_frm_ready = PJ_TRUE;
+    while (!play_frm_copied) {
+	pj_thread_sleep(1);
+    }
+    play_frm_ready = PJ_FALSE;
+
+    return PJ_SUCCESS;
+}
+
+static pj_status_t play_cb(void *user_data, pjmedia_frame *f)
+{
+    PJ_UNUSED_ARG(user_data);
+
+    if (!play_frm_ready) {
+	PJ_LOG(3, ("play_cb()", "Warning! Play frame not ready")); 
+	return PJ_SUCCESS;
+    }
+
+    pj_memcpy(f->buf, play_frm.buf, play_frm.size);
+    f->size = play_frm.size;
+
+    play_frm_copied = PJ_TRUE;
+    return PJ_SUCCESS;
+}
+
+static void pcap2wav(const pj_str_t *codec,
+		     const pj_str_t *wav_filename,
+		     pjmedia_aud_dev_index dev_id,
+		     const pj_str_t *srtp_crypto,
+		     const pj_str_t *srtp_key)
+{
+    const pj_str_t WAV = {".wav", 4};
+    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 */
+    T( pjmedia_codec_register_audio_codecs(app.mept, NULL) );
+
+    /* 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, PJ_FALSE);
+
+    cmgr = pjmedia_endpt_get_codec_mgr(app.mept);
+
+    /* Get codec info and param for the specified payload type */
+    app.pt = pkt0.rtp->pt;
+    if (app.pt >=0 && app.pt < 96) {
+	T( pjmedia_codec_mgr_get_codec_info(cmgr, pkt0.rtp->pt, &ci) );
+    } else {
+	unsigned cnt = 2;
+	const pjmedia_codec_info *info[2];
+	T( pjmedia_codec_mgr_find_codecs_by_id(cmgr, codec, &cnt, 
+					       info, NULL) );
+	if (cnt != 1)
+	    err_exit("Codec ID must be specified and unique!", 0);
+
+	ci = info[0];
+    }
+    T( pjmedia_codec_mgr_get_default_param(cmgr, ci, &param) );
+
+    /* Alloc and init codec */
+    T( pjmedia_codec_mgr_alloc_codec(cmgr, ci, &app.codec) );
+    T( pjmedia_codec_init(app.codec, app.pool) );
+    T( pjmedia_codec_open(app.codec, &param) );
+
+    /* Init audio device or WAV file */
+    samples_per_frame = ci->clock_rate * param.info.frm_ptime / 1000;
+    if (pj_strcmp2(wav_filename, "-") == 0) {
+	pjmedia_aud_param aud_param;
+
+	/* Open audio device */
+	T( pjmedia_aud_dev_default_param(dev_id, &aud_param) );
+	aud_param.dir = PJMEDIA_DIR_PLAYBACK;
+	aud_param.channel_count = ci->channel_cnt;
+	aud_param.clock_rate = ci->clock_rate;
+	aud_param.samples_per_frame = samples_per_frame;
+	T( pjmedia_aud_stream_create(&aud_param, NULL, &play_cb, 
+				     NULL, &app.aud_strm) );
+	T( pjmedia_aud_stream_start(app.aud_strm) );
+    } else if (pj_stristr(wav_filename, &WAV)) {
+	/* Open WAV file */
+	T( pjmedia_wav_writer_port_create(app.pool, wav_filename->ptr,
+					  ci->clock_rate, ci->channel_cnt,
+					  samples_per_frame,
+					  param.info.pcm_bits_per_sample, 0, 0,
+					  &app.wav) );
+    } else {
+	err_exit("invalid output file", PJ_EINVAL);
+    }
+
+    /* 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( pjmedia_codec_parse(app.codec, pkt0.payload, pkt0.payload_len, 
+				&ts, &frame_cnt, frames) );
+
+	/* Decode and write to WAV file */
+	samples_cnt = 0;
+	for (i=0; i<frame_cnt; ++i) {
+	    pjmedia_frame pcm_frame;
+
+	    pcm_frame.buf = pcm;
+	    pcm_frame.size = samples_per_frame * 2;
+
+	    T( pjmedia_codec_decode(app.codec, &frames[i], 
+				    (unsigned)pcm_frame.size, &pcm_frame) );
+	    if (app.wav) {
+		T( pjmedia_port_put_frame(app.wav, &pcm_frame) );
+	    }
+	    if (app.aud_strm) {
+		T( wait_play(&pcm_frame) );
+	    }
+	    samples_cnt += samples_per_frame;
+	}
+
+	/* Read next packet */
+	read_rtp(pkt1.buffer, sizeof(pkt1.buffer), &pkt1.rtp,
+		 &pkt1.payload, &pkt1.payload_len, PJ_TRUE);
+
+	/* 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( pjmedia_codec_recover(app.codec, (unsigned)pcm_frame.size, 
+					 &pcm_frame) );
+	    } else {
+		pj_bzero(pcm_frame.buf, pcm_frame.size);
+	    }
+
+	    if (app.wav) {
+		T( pjmedia_port_put_frame(app.wav, &pcm_frame) );
+	    }
+	    if (app.aud_strm) {
+		T( wait_play(&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, srtp_crypto, srtp_key, codec;
+    pjmedia_aud_dev_index dev_id = PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV;
+    pj_pcap_filter filter;
+    pj_status_t status;
+
+    enum { 
+	OPT_SRC_IP = 1, OPT_DST_IP, OPT_SRC_PORT, OPT_DST_PORT,
+	OPT_CODEC, OPT_PLAY_DEV_ID
+    };
+    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 },
+	{ "codec",	    1, 0, OPT_CODEC },
+	{ "play-dev-id",    1, 0, OPT_PLAY_DEV_ID },
+	{ NULL, 0, 0, 0}
+    };
+    int c;
+    int option_index;
+    char key_bin[32];
+
+    srtp_crypto.slen = srtp_key.slen = 0;
+    codec.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;
+	case OPT_CODEC:
+	    codec = pj_str(pj_optarg);
+	    break;
+	case OPT_PLAY_DEV_ID:
+	    dev_id = 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]);
+    
+    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) );
+
+    pcap2wav(&codec, &output, dev_id, &srtp_crypto, &srtp_key);
+
+    cleanup();
+    return 0;
+}
+