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-rw-r--r--pjmedia/src/pjmedia/stream_info.c548
1 files changed, 548 insertions, 0 deletions
diff --git a/pjmedia/src/pjmedia/stream_info.c b/pjmedia/src/pjmedia/stream_info.c
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+++ b/pjmedia/src/pjmedia/stream_info.c
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+/* $Id$ */
+/*
+ * 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 <pjmedia/stream.h>
+#include <pjmedia/stream_common.h>
+#include <pj/ctype.h>
+#include <pj/rand.h>
+
+static const pj_str_t ID_AUDIO = { "audio", 5};
+static const pj_str_t ID_IN = { "IN", 2 };
+static const pj_str_t ID_IP4 = { "IP4", 3};
+static const pj_str_t ID_IP6 = { "IP6", 3};
+static const pj_str_t ID_RTP_AVP = { "RTP/AVP", 7 };
+static const pj_str_t ID_RTP_SAVP = { "RTP/SAVP", 8 };
+//static const pj_str_t ID_SDP_NAME = { "pjmedia", 7 };
+static const pj_str_t ID_RTPMAP = { "rtpmap", 6 };
+static const pj_str_t ID_TELEPHONE_EVENT = { "telephone-event", 15 };
+
+static const pj_str_t STR_INACTIVE = { "inactive", 8 };
+static const pj_str_t STR_SENDRECV = { "sendrecv", 8 };
+static const pj_str_t STR_SENDONLY = { "sendonly", 8 };
+static const pj_str_t STR_RECVONLY = { "recvonly", 8 };
+
+
+/*
+ * Internal function for collecting codec info and param from the SDP media.
+ */
+static pj_status_t get_audio_codec_info_param(pjmedia_stream_info *si,
+ pj_pool_t *pool,
+ pjmedia_codec_mgr *mgr,
+ const pjmedia_sdp_media *local_m,
+ const pjmedia_sdp_media *rem_m)
+{
+ const pjmedia_sdp_attr *attr;
+ pjmedia_sdp_rtpmap *rtpmap;
+ unsigned i, fmti, pt = 0;
+ pj_status_t status;
+
+ /* Find the first codec which is not telephone-event */
+ for ( fmti = 0; fmti < local_m->desc.fmt_count; ++fmti ) {
+ pjmedia_sdp_rtpmap r;
+
+ if ( !pj_isdigit(*local_m->desc.fmt[fmti].ptr) )
+ return PJMEDIA_EINVALIDPT;
+ pt = pj_strtoul(&local_m->desc.fmt[fmti]);
+
+ if (pt < 96) {
+ /* This is known static PT. Skip rtpmap checking because it is
+ * optional. */
+ break;
+ }
+
+ attr = pjmedia_sdp_media_find_attr(local_m, &ID_RTPMAP,
+ &local_m->desc.fmt[fmti]);
+ if (attr == NULL)
+ continue;
+
+ status = pjmedia_sdp_attr_get_rtpmap(attr, &r);
+ if (status != PJ_SUCCESS)
+ continue;
+
+ if (pj_strcmp(&r.enc_name, &ID_TELEPHONE_EVENT) != 0)
+ break;
+ }
+ if ( fmti >= local_m->desc.fmt_count )
+ return PJMEDIA_EINVALIDPT;
+
+ /* Get payload type for receiving direction */
+ si->rx_pt = pt;
+
+ /* Get codec info.
+ * For static payload types, get the info from codec manager.
+ * For dynamic payload types, MUST get the rtpmap.
+ */
+ if (pt < 96) {
+ pj_bool_t has_rtpmap;
+
+ rtpmap = NULL;
+ has_rtpmap = PJ_TRUE;
+
+ attr = pjmedia_sdp_media_find_attr(local_m, &ID_RTPMAP,
+ &local_m->desc.fmt[fmti]);
+ if (attr == NULL) {
+ has_rtpmap = PJ_FALSE;
+ }
+ if (attr != NULL) {
+ status = pjmedia_sdp_attr_to_rtpmap(pool, attr, &rtpmap);
+ if (status != PJ_SUCCESS)
+ has_rtpmap = PJ_FALSE;
+ }
+
+ /* Build codec format info: */
+ if (has_rtpmap) {
+ si->fmt.type = si->type;
+ si->fmt.pt = pj_strtoul(&local_m->desc.fmt[fmti]);
+ pj_strdup(pool, &si->fmt.encoding_name, &rtpmap->enc_name);
+ si->fmt.clock_rate = rtpmap->clock_rate;
+
+#if defined(PJMEDIA_HANDLE_G722_MPEG_BUG) && (PJMEDIA_HANDLE_G722_MPEG_BUG != 0)
+ /* The session info should have the actual clock rate, because
+ * this info is used for calculationg buffer size, etc in stream
+ */
+ if (si->fmt.pt == PJMEDIA_RTP_PT_G722)
+ si->fmt.clock_rate = 16000;
+#endif
+
+ /* For audio codecs, rtpmap parameters denotes the number of
+ * channels.
+ */
+ if (si->type == PJMEDIA_TYPE_AUDIO && rtpmap->param.slen) {
+ si->fmt.channel_cnt = (unsigned) pj_strtoul(&rtpmap->param);
+ } else {
+ si->fmt.channel_cnt = 1;
+ }
+
+ } else {
+ const pjmedia_codec_info *p_info;
+
+ status = pjmedia_codec_mgr_get_codec_info( mgr, pt, &p_info);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ pj_memcpy(&si->fmt, p_info, sizeof(pjmedia_codec_info));
+ }
+
+ /* For static payload type, pt's are symetric */
+ si->tx_pt = pt;
+
+ } else {
+ pjmedia_codec_id codec_id;
+ pj_str_t codec_id_st;
+ const pjmedia_codec_info *p_info;
+
+ attr = pjmedia_sdp_media_find_attr(local_m, &ID_RTPMAP,
+ &local_m->desc.fmt[fmti]);
+ if (attr == NULL)
+ return PJMEDIA_EMISSINGRTPMAP;
+
+ status = pjmedia_sdp_attr_to_rtpmap(pool, attr, &rtpmap);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ /* Build codec format info: */
+
+ si->fmt.type = si->type;
+ si->fmt.pt = pj_strtoul(&local_m->desc.fmt[fmti]);
+ si->fmt.encoding_name = rtpmap->enc_name;
+ si->fmt.clock_rate = rtpmap->clock_rate;
+
+ /* For audio codecs, rtpmap parameters denotes the number of
+ * channels.
+ */
+ if (si->type == PJMEDIA_TYPE_AUDIO && rtpmap->param.slen) {
+ si->fmt.channel_cnt = (unsigned) pj_strtoul(&rtpmap->param);
+ } else {
+ si->fmt.channel_cnt = 1;
+ }
+
+ /* Normalize the codec info from codec manager. Note that the
+ * payload type will be resetted to its default (it might have
+ * been rewritten by the SDP negotiator to match to the remote
+ * offer), this is intentional as currently some components may
+ * prefer (or even require) the default PT in codec info.
+ */
+ pjmedia_codec_info_to_id(&si->fmt, codec_id, sizeof(codec_id));
+
+ i = 1;
+ codec_id_st = pj_str(codec_id);
+ status = pjmedia_codec_mgr_find_codecs_by_id(mgr, &codec_id_st,
+ &i, &p_info, NULL);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ pj_memcpy(&si->fmt, p_info, sizeof(pjmedia_codec_info));
+
+ /* Determine payload type for outgoing channel, by finding
+ * dynamic payload type in remote SDP that matches the answer.
+ */
+ si->tx_pt = 0xFFFF;
+ for (i=0; i<rem_m->desc.fmt_count; ++i) {
+ unsigned rpt;
+ pjmedia_sdp_attr *r_attr;
+ pjmedia_sdp_rtpmap r_rtpmap;
+
+ rpt = pj_strtoul(&rem_m->desc.fmt[i]);
+ if (rpt < 96)
+ continue;
+
+ r_attr = pjmedia_sdp_media_find_attr(rem_m, &ID_RTPMAP,
+ &rem_m->desc.fmt[i]);
+ if (!r_attr)
+ continue;
+
+ if (pjmedia_sdp_attr_get_rtpmap(r_attr, &r_rtpmap) != PJ_SUCCESS)
+ continue;
+
+ if (!pj_stricmp(&rtpmap->enc_name, &r_rtpmap.enc_name) &&
+ rtpmap->clock_rate == r_rtpmap.clock_rate)
+ {
+ /* Found matched codec. */
+ si->tx_pt = rpt;
+
+ break;
+ }
+ }
+
+ if (si->tx_pt == 0xFFFF)
+ return PJMEDIA_EMISSINGRTPMAP;
+ }
+
+
+ /* Now that we have codec info, get the codec param. */
+ si->param = PJ_POOL_ALLOC_T(pool, pjmedia_codec_param);
+ status = pjmedia_codec_mgr_get_default_param(mgr, &si->fmt,
+ si->param);
+
+ /* Get remote fmtp for our encoder. */
+ pjmedia_stream_info_parse_fmtp(pool, rem_m, si->tx_pt,
+ &si->param->setting.enc_fmtp);
+
+ /* Get local fmtp for our decoder. */
+ pjmedia_stream_info_parse_fmtp(pool, local_m, si->rx_pt,
+ &si->param->setting.dec_fmtp);
+
+ /* Get the remote ptime for our encoder. */
+ attr = pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
+ "ptime", NULL);
+ if (attr) {
+ pj_str_t tmp_val = attr->value;
+ unsigned frm_per_pkt;
+
+ pj_strltrim(&tmp_val);
+
+ /* Round up ptime when the specified is not multiple of frm_ptime */
+ frm_per_pkt = (pj_strtoul(&tmp_val) +
+ si->param->info.frm_ptime/2) /
+ si->param->info.frm_ptime;
+ if (frm_per_pkt != 0) {
+ si->param->setting.frm_per_pkt = (pj_uint8_t)frm_per_pkt;
+ }
+ }
+
+ /* Get remote maxptime for our encoder. */
+ attr = pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
+ "maxptime", NULL);
+ if (attr) {
+ pj_str_t tmp_val = attr->value;
+
+ pj_strltrim(&tmp_val);
+ si->tx_maxptime = pj_strtoul(&tmp_val);
+ }
+
+ /* When direction is NONE (it means SDP negotiation has failed) we don't
+ * need to return a failure here, as returning failure will cause
+ * the whole SDP to be rejected. See ticket #:
+ * http://
+ *
+ * Thanks Alain Totouom
+ */
+ if (status != PJ_SUCCESS && si->dir != PJMEDIA_DIR_NONE)
+ return status;
+
+
+ /* Get incomming payload type for telephone-events */
+ si->rx_event_pt = -1;
+ for (i=0; i<local_m->attr_count; ++i) {
+ pjmedia_sdp_rtpmap r;
+
+ attr = local_m->attr[i];
+ if (pj_strcmp(&attr->name, &ID_RTPMAP) != 0)
+ continue;
+ if (pjmedia_sdp_attr_get_rtpmap(attr, &r) != PJ_SUCCESS)
+ continue;
+ if (pj_strcmp(&r.enc_name, &ID_TELEPHONE_EVENT) == 0) {
+ si->rx_event_pt = pj_strtoul(&r.pt);
+ break;
+ }
+ }
+
+ /* Get outgoing payload type for telephone-events */
+ si->tx_event_pt = -1;
+ for (i=0; i<rem_m->attr_count; ++i) {
+ pjmedia_sdp_rtpmap r;
+
+ attr = rem_m->attr[i];
+ if (pj_strcmp(&attr->name, &ID_RTPMAP) != 0)
+ continue;
+ if (pjmedia_sdp_attr_get_rtpmap(attr, &r) != PJ_SUCCESS)
+ continue;
+ if (pj_strcmp(&r.enc_name, &ID_TELEPHONE_EVENT) == 0) {
+ si->tx_event_pt = pj_strtoul(&r.pt);
+ break;
+ }
+ }
+
+ return PJ_SUCCESS;
+}
+
+
+
+/*
+ * Create stream info from SDP media line.
+ */
+PJ_DEF(pj_status_t) pjmedia_stream_info_from_sdp(
+ pjmedia_stream_info *si,
+ pj_pool_t *pool,
+ pjmedia_endpt *endpt,
+ const pjmedia_sdp_session *local,
+ const pjmedia_sdp_session *remote,
+ unsigned stream_idx)
+{
+ pjmedia_codec_mgr *mgr;
+ const pjmedia_sdp_attr *attr;
+ const pjmedia_sdp_media *local_m;
+ const pjmedia_sdp_media *rem_m;
+ const pjmedia_sdp_conn *local_conn;
+ const pjmedia_sdp_conn *rem_conn;
+ int rem_af, local_af;
+ pj_sockaddr local_addr;
+ pj_status_t status;
+
+
+ /* Validate arguments: */
+ PJ_ASSERT_RETURN(pool && si && local && remote, PJ_EINVAL);
+ PJ_ASSERT_RETURN(stream_idx < local->media_count, PJ_EINVAL);
+ PJ_ASSERT_RETURN(stream_idx < remote->media_count, PJ_EINVAL);
+
+ /* Keep SDP shortcuts */
+ local_m = local->media[stream_idx];
+ rem_m = remote->media[stream_idx];
+
+ local_conn = local_m->conn ? local_m->conn : local->conn;
+ if (local_conn == NULL)
+ return PJMEDIA_SDP_EMISSINGCONN;
+
+ rem_conn = rem_m->conn ? rem_m->conn : remote->conn;
+ if (rem_conn == NULL)
+ return PJMEDIA_SDP_EMISSINGCONN;
+
+ /* Media type must be audio */
+ if (pj_stricmp(&local_m->desc.media, &ID_AUDIO) != 0)
+ return PJMEDIA_EINVALIMEDIATYPE;
+
+ /* Get codec manager. */
+ mgr = pjmedia_endpt_get_codec_mgr(endpt);
+
+ /* Reset: */
+
+ pj_bzero(si, sizeof(*si));
+
+#if PJMEDIA_HAS_RTCP_XR && PJMEDIA_STREAM_ENABLE_XR
+ /* Set default RTCP XR enabled/disabled */
+ si->rtcp_xr_enabled = PJ_TRUE;
+#endif
+
+ /* Media type: */
+ si->type = PJMEDIA_TYPE_AUDIO;
+
+ /* Transport protocol */
+
+ /* At this point, transport type must be compatible,
+ * the transport instance will do more validation later.
+ */
+ status = pjmedia_sdp_transport_cmp(&rem_m->desc.transport,
+ &local_m->desc.transport);
+ if (status != PJ_SUCCESS)
+ return PJMEDIA_SDPNEG_EINVANSTP;
+
+ if (pj_stricmp(&local_m->desc.transport, &ID_RTP_AVP) == 0) {
+
+ si->proto = PJMEDIA_TP_PROTO_RTP_AVP;
+
+ } else if (pj_stricmp(&local_m->desc.transport, &ID_RTP_SAVP) == 0) {
+
+ si->proto = PJMEDIA_TP_PROTO_RTP_SAVP;
+
+ } else {
+
+ si->proto = PJMEDIA_TP_PROTO_UNKNOWN;
+ return PJ_SUCCESS;
+ }
+
+
+ /* Check address family in remote SDP */
+ rem_af = pj_AF_UNSPEC();
+ if (pj_stricmp(&rem_conn->net_type, &ID_IN)==0) {
+ if (pj_stricmp(&rem_conn->addr_type, &ID_IP4)==0) {
+ rem_af = pj_AF_INET();
+ } else if (pj_stricmp(&rem_conn->addr_type, &ID_IP6)==0) {
+ rem_af = pj_AF_INET6();
+ }
+ }
+
+ if (rem_af==pj_AF_UNSPEC()) {
+ /* Unsupported address family */
+ return PJ_EAFNOTSUP;
+ }
+
+ /* Set remote address: */
+ status = pj_sockaddr_init(rem_af, &si->rem_addr, &rem_conn->addr,
+ rem_m->desc.port);
+ if (status != PJ_SUCCESS) {
+ /* Invalid IP address. */
+ return PJMEDIA_EINVALIDIP;
+ }
+
+ /* Check address family of local info */
+ local_af = pj_AF_UNSPEC();
+ if (pj_stricmp(&local_conn->net_type, &ID_IN)==0) {
+ if (pj_stricmp(&local_conn->addr_type, &ID_IP4)==0) {
+ local_af = pj_AF_INET();
+ } else if (pj_stricmp(&local_conn->addr_type, &ID_IP6)==0) {
+ local_af = pj_AF_INET6();
+ }
+ }
+
+ if (local_af==pj_AF_UNSPEC()) {
+ /* Unsupported address family */
+ return PJ_SUCCESS;
+ }
+
+ /* Set remote address: */
+ status = pj_sockaddr_init(local_af, &local_addr, &local_conn->addr,
+ local_m->desc.port);
+ if (status != PJ_SUCCESS) {
+ /* Invalid IP address. */
+ return PJMEDIA_EINVALIDIP;
+ }
+
+ /* Local and remote address family must match */
+ if (local_af != rem_af)
+ return PJ_EAFNOTSUP;
+
+ /* Media direction: */
+
+ if (local_m->desc.port == 0 ||
+ pj_sockaddr_has_addr(&local_addr)==PJ_FALSE ||
+ pj_sockaddr_has_addr(&si->rem_addr)==PJ_FALSE ||
+ pjmedia_sdp_media_find_attr(local_m, &STR_INACTIVE, NULL)!=NULL)
+ {
+ /* Inactive stream. */
+
+ si->dir = PJMEDIA_DIR_NONE;
+
+ } else if (pjmedia_sdp_media_find_attr(local_m, &STR_SENDONLY, NULL)!=NULL) {
+
+ /* Send only stream. */
+
+ si->dir = PJMEDIA_DIR_ENCODING;
+
+ } else if (pjmedia_sdp_media_find_attr(local_m, &STR_RECVONLY, NULL)!=NULL) {
+
+ /* Recv only stream. */
+
+ si->dir = PJMEDIA_DIR_DECODING;
+
+ } else {
+
+ /* Send and receive stream. */
+
+ si->dir = PJMEDIA_DIR_ENCODING_DECODING;
+
+ }
+
+ /* No need to do anything else if stream is rejected */
+ if (local_m->desc.port == 0) {
+ return PJ_SUCCESS;
+ }
+
+ /* If "rtcp" attribute is present in the SDP, set the RTCP address
+ * from that attribute. Otherwise, calculate from RTP address.
+ */
+ attr = pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
+ "rtcp", NULL);
+ if (attr) {
+ pjmedia_sdp_rtcp_attr rtcp;
+ status = pjmedia_sdp_attr_get_rtcp(attr, &rtcp);
+ if (status == PJ_SUCCESS) {
+ if (rtcp.addr.slen) {
+ status = pj_sockaddr_init(rem_af, &si->rem_rtcp, &rtcp.addr,
+ (pj_uint16_t)rtcp.port);
+ } else {
+ pj_sockaddr_init(rem_af, &si->rem_rtcp, NULL,
+ (pj_uint16_t)rtcp.port);
+ pj_memcpy(pj_sockaddr_get_addr(&si->rem_rtcp),
+ pj_sockaddr_get_addr(&si->rem_addr),
+ pj_sockaddr_get_addr_len(&si->rem_addr));
+ }
+ }
+ }
+
+ if (!pj_sockaddr_has_addr(&si->rem_rtcp)) {
+ int rtcp_port;
+
+ pj_memcpy(&si->rem_rtcp, &si->rem_addr, sizeof(pj_sockaddr));
+ rtcp_port = pj_sockaddr_get_port(&si->rem_addr) + 1;
+ pj_sockaddr_set_port(&si->rem_rtcp, (pj_uint16_t)rtcp_port);
+ }
+
+
+ /* Get the payload number for receive channel. */
+ /*
+ Previously we used to rely on fmt[0] being the selected codec,
+ but some UA sends telephone-event as fmt[0] and this would
+ cause assert failure below.
+
+ Thanks Chris Hamilton <chamilton .at. cs.dal.ca> for this patch.
+
+ // And codec must be numeric!
+ if (!pj_isdigit(*local_m->desc.fmt[0].ptr) ||
+ !pj_isdigit(*rem_m->desc.fmt[0].ptr))
+ {
+ return PJMEDIA_EINVALIDPT;
+ }
+
+ pt = pj_strtoul(&local_m->desc.fmt[0]);
+ pj_assert(PJMEDIA_RTP_PT_TELEPHONE_EVENTS==0 ||
+ pt != PJMEDIA_RTP_PT_TELEPHONE_EVENTS);
+ */
+
+ /* Get codec info and param */
+ status = get_audio_codec_info_param(si, pool, mgr, local_m, rem_m);
+
+ /* Leave SSRC to random. */
+ si->ssrc = pj_rand();
+
+ /* Set default jitter buffer parameter. */
+ si->jb_init = si->jb_max = si->jb_min_pre = si->jb_max_pre = -1;
+
+ return status;
+}
+
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-25 22:33:42 +0000