/* $Id: stream_info.c 3982 2012-03-22 09:56:52Z bennylp $ */ /* * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) * Copyright (C) 2003-2008 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 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; idesc.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; iattr_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; iattr_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 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; }