/* * (C) 2009-2015 by Holger Hans Peter Freyther * (C) 2009-2014 by On-Waves * All Rights Reserved * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU Affero General Public License as published by * the Free Software Foundation; either version 3 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 Affero General Public License for more details. * * You should have received a copy of the GNU Affero General Public License * along with this program. If not, see . * */ #include #include #include #include #include #include #include #include /* Helper function to dump codec information of a specified codec to a printable * string, used by dump_codec_summary() */ static char *dump_codec(struct mgcp_rtp_codec *codec) { static char str[256]; char *pt_str; if (codec->payload_type > 76) pt_str = "DYNAMIC"; else if (codec->payload_type > 72) pt_str = "RESERVED "; else if (codec->payload_type != PTYPE_UNDEFINED) pt_str = codec->subtype_name; else pt_str = "INVALID "; snprintf(str, sizeof(str), "(pt:%i=%s, audio:%s subt=%s, rate=%u, ch=%i, t=%u/%u)", codec->payload_type, pt_str, codec->audio_name, codec->subtype_name, codec->rate, codec->channels, codec->frame_duration_num, codec->frame_duration_den); return str; } /*! Dump a summary of all negotiated codecs to debug log * \param[in] conn related rtp-connection. */ void mgcp_codec_summary(struct mgcp_conn_rtp *conn) { struct mgcp_rtp_end *rtp; unsigned int i; struct mgcp_rtp_codec *codec; struct mgcp_endpoint *endp; rtp = &conn->end; endp = conn->conn->endp; if (rtp->codecs_assigned == 0) { LOGPENDP(endp, DLMGCP, LOGL_ERROR, "conn:%s no codecs available\n", mgcp_conn_dump(conn->conn)); return; } /* Store parsed codec information */ for (i = 0; i < rtp->codecs_assigned; i++) { codec = &rtp->codecs[i]; LOGPENDP(endp, DLMGCP, LOGL_DEBUG, "conn:%s codecs[%u]:%s", mgcp_conn_dump(conn->conn), i, dump_codec(codec)); if (codec == rtp->codec) LOGPC(DLMGCP, LOGL_DEBUG, " [selected]"); LOGPC(DLMGCP, LOGL_DEBUG, "\n"); } } /* Initalize or reset codec information with default data. */ static void codec_init(struct mgcp_rtp_codec *codec) { *codec = (struct mgcp_rtp_codec){ .payload_type = -1, .frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM, .frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN, .rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE, .channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS, .subtype_name = "", .audio_name = "", }; } static void codec_free(struct mgcp_rtp_codec *codec) { *codec = (struct mgcp_rtp_codec){}; } /*! Initalize or reset codec information with default data. * \param[out] conn related rtp-connection. */ void mgcp_codec_reset_all(struct mgcp_conn_rtp *conn) { int i; for (i = 0; i < conn->end.codecs_assigned; i++) codec_free(&conn->end.codecs[i]); conn->end.codecs_assigned = 0; conn->end.codec = NULL; } /*! Add codec configuration depending on payload type and/or codec name. This * function uses the input parameters to extrapolate the full codec information. * \param[out] codec configuration (caller provided memory). * \param[out] conn related rtp-connection. * \param[in] payload_type codec type id (e.g. 3 for GSM, -1 when undefined). * \param[in] audio_name audio codec name, in uppercase (e.g. "GSM/8000/1"). * \param[in] param optional codec parameters (set to NULL when unused). * \returns 0 on success, -EINVAL on failure. */ int mgcp_codec_add(struct mgcp_conn_rtp *conn, int payload_type, const char *audio_name, const struct mgcp_codec_param *param) { int rate; int channels; struct mgcp_rtp_codec *codec; unsigned int pt_offset = conn->end.codecs_assigned; /* The amount of codecs we can store is limited, make sure we do not * overrun this limit. */ if (conn->end.codecs_assigned >= MGCP_MAX_CODECS) return -EINVAL; /* First unused entry */ codec = &conn->end.codecs[conn->end.codecs_assigned]; /* Initalize the codec struct with some default data to begin with */ codec_init(codec); if (payload_type != PTYPE_UNDEFINED) { /* Make sure we do not get any reserved or undefined type numbers */ /* See also: https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml */ if ((payload_type == 1 || payload_type == 2 || payload_type == 19) || (payload_type >= 72 && payload_type <= 76) || (payload_type >= 127)) { LOGP(DLMGCP, LOGL_ERROR, "Cannot add codec, payload type number %d is reserved\n", payload_type); goto error; } codec->payload_type = payload_type; } /* When no audio name is given, we are forced to use the payload * type to generate the audio name. This is only possible for * non dynamic payload types, which are statically defined */ if (!audio_name) { switch (payload_type) { case 0: strcpy(codec->audio_name, "PCMU/8000/1"); break; case 3: strcpy(codec->audio_name, "GSM/8000/1"); break; case 8: strcpy(codec->audio_name, "PCMA/8000/1"); break; case 18: strcpy(codec->audio_name, "G729/8000/1"); break; default: /* The given payload type is not known to us, or it * it is a dynamic payload type for which we do not * know the audio name. We must give up here */ LOGP(DLMGCP, LOGL_ERROR, "No audio codec name given, and payload type %d unknown\n", payload_type); goto error; } } else { OSMO_STRLCPY_ARRAY(codec->audio_name, audio_name); } /* Now we extract the codec subtype name, rate and channels. The latter * two are optional. If they are not present we use the safe defaults * above. */ if (strlen(codec->audio_name) >= sizeof(codec->subtype_name)) { LOGP(DLMGCP, LOGL_ERROR, "Audio codec too long: %s\n", osmo_quote_str(codec->audio_name, -1)); goto error; } channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS; rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE; if (sscanf(codec->audio_name, "%63[^/]/%d/%d", codec->subtype_name, &rate, &channels) < 1) { LOGP(DLMGCP, LOGL_ERROR, "Invalid audio codec: %s\n", osmo_quote_str(codec->audio_name, -1)); goto error; } /* Note: We only accept configurations with one audio channel! */ if (channels != 1) { LOGP(DLMGCP, LOGL_ERROR, "Cannot handle audio codec with more than one channel: %s\n", osmo_quote_str(codec->audio_name, -1)); goto error; } codec->rate = rate; codec->channels = channels; codec->payload_type = payload_type; if (!strcmp(codec->subtype_name, "G729")) { codec->frame_duration_num = 10; codec->frame_duration_den = 1000; } else { codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM; codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN; } /* Derive the payload type if it is unknown */ if (codec->payload_type == PTYPE_UNDEFINED) { /* TODO: This is semi dead code, see OS#4150 */ /* For the known codecs from the static range we restore * the IANA or 3GPP assigned payload type number */ if (codec->rate == 8000 && codec->channels == 1) { /* See also: https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml */ if (!strcmp(codec->subtype_name, "GSM")) codec->payload_type = 3; else if (!strcmp(codec->subtype_name, "PCMA")) codec->payload_type = 8; else if (!strcmp(codec->subtype_name, "PCMU")) codec->payload_type = 0; else if (!strcmp(codec->subtype_name, "G729")) codec->payload_type = 18; /* See also: 3GPP TS 48.103, chapter 5.4.2.2 RTP Payload * Note: These are not fixed payload types as the IANA * defined once, they still remain dymanic payload * types, but with a payload type number preference. */ else if (!strcmp(codec->subtype_name, "GSM-EFR")) codec->payload_type = 110; else if (!strcmp(codec->subtype_name, "GSM-HR-08")) codec->payload_type = 111; else if (!strcmp(codec->subtype_name, "AMR")) codec->payload_type = 112; else if (!strcmp(codec->subtype_name, "AMR-WB")) codec->payload_type = 113; } /* If we could not determine a payload type we assume that * we are dealing with a codec from the dynamic range. We * choose a fixed identifier from 96-109. (Note: normally, * the dynamic payload type rante is from 96-127, but from * 110 onwards 3gpp defines prefered codec types, which are * also fixed, see above) */ if (codec->payload_type < 0) { /* FIXME: pt_offset is completely unrelated and useless here, any of those numbers may already * have been added to the codecs. Instead, there should be an iterator checking for an actually * unused dynamic payload type number. */ codec->payload_type = 96 + pt_offset; if (codec->payload_type > 109) { LOGP(DLMGCP, LOGL_ERROR, "Ran out of payload type numbers to assign dynamically\n"); goto error; } } } /* Copy over optional codec parameters */ if (param) { codec->param = *param; codec->param_present = true; } else codec->param_present = false; conn->end.codecs_assigned++; return 0; error: /* Make sure we leave a clean codec entry on error. */ codec_free(codec); return -EINVAL; } /* Return true if octet-aligned is set in the given codec. Default to octet-aligned=0, i.e. bandwidth-efficient mode. * See RFC4867 "RTP Payload Format for AMR and AMR-WB" sections "8.1. AMR Media Type Registration" and "8.2. AMR-WB * Media Type Registration": * * octet-align: Permissible values are 0 and 1. If 1, octet-aligned * operation SHALL be used. If 0 or if not present, * bandwidth-efficient operation is employed. * * https://tools.ietf.org/html/rfc4867 */ bool mgcp_codec_amr_is_octet_aligned(const struct mgcp_rtp_codec *codec) { if (!codec->param_present) return false; if (!codec->param.amr_octet_aligned_present) return false; return codec->param.amr_octet_aligned; } /* Compare two codecs, all parameters must match up */ static bool codecs_same(struct mgcp_rtp_codec *codec_a, struct mgcp_rtp_codec *codec_b) { /* All codec properties must match up, except the payload type number. Even though standardisd payload numbers * exist for certain situations, the call agent may still assign them freely. Hence we must not insist on equal * payload type numbers. Also the audio_name is not checked since it is already parsed into subtype_name, rate, * and channels, which are checked. */ if (strcmp(codec_a->subtype_name, codec_b->subtype_name)) return false; if (codec_a->rate != codec_b->rate) return false; if (codec_a->channels != codec_b->channels) return false; if (codec_a->frame_duration_num != codec_b->frame_duration_num) return false; if (codec_a->frame_duration_den != codec_b->frame_duration_den) return false; /* AMR payload may be formatted in two different payload formats, it is still the same codec but since the * formatting of the payload is different, conversation is required, so we must treat it as a different * codec here. */ if (strcmp(codec_a->subtype_name, "AMR") == 0) { if (mgcp_codec_amr_is_octet_aligned(codec_a) != mgcp_codec_amr_is_octet_aligned(codec_b)) return false; } return true; } /* Compare two codecs, all parameters must match up, except parameters related to payload formatting (not checked). */ static bool codecs_convertible(struct mgcp_rtp_codec *codec_a, struct mgcp_rtp_codec *codec_b) { /* OsmoMGW currently has no ability to transcode from one codec to another. However OsmoMGW is still able to * translate between different payload formats as long as the encoded voice data itself does not change. * Therefore we must insist on equal codecs but still allow different payload formatting. */ /* In 3G IuUP, AMR may be encapsulated in IuFP, this means even though the codec name and negotiated rate is * different, the formatting can still be converted by OsmoMGW. Therefore we won't insist on equal * subtype_name and rate if we detect IuFP and AMR is used on the same tandem. */ if (strcmp(codec_a->subtype_name, "AMR") == 0 && strcmp(codec_b->subtype_name, "VND.3GPP.IUFP") == 0) goto iufp; if (strcmp(codec_a->subtype_name, "VND.3GPP.IUFP") == 0 && strcmp(codec_b->subtype_name, "AMR") == 0) goto iufp; if (strcmp(codec_a->subtype_name, codec_b->subtype_name)) return false; if (codec_a->rate != codec_b->rate) return false; iufp: if (codec_a->channels != codec_b->channels) return false; if (codec_a->frame_duration_num != codec_b->frame_duration_num) return false; if (codec_a->frame_duration_den != codec_b->frame_duration_den) return false; return true; } struct mgcp_rtp_codec *mgcp_codec_find_same(struct mgcp_conn_rtp *conn, struct mgcp_rtp_codec *codec) { struct mgcp_rtp_end *rtp_end; unsigned int i; unsigned int codecs_assigned; rtp_end = &conn->end; /* Use the codec information from the source and try to find the equivalent of it on the destination side. In * the first run we will look for an exact match. */ codecs_assigned = rtp_end->codecs_assigned; OSMO_ASSERT(codecs_assigned <= MGCP_MAX_CODECS); for (i = 0; i < codecs_assigned; i++) { if (codecs_same(codec, &rtp_end->codecs[i])) { return &rtp_end->codecs[i]; break; } } return NULL; } /* For a given codec, find a convertible codec in the given connection. */ static struct mgcp_rtp_codec *codec_find_convertible(struct mgcp_conn_rtp *conn, struct mgcp_rtp_codec *codec) { struct mgcp_rtp_end *rtp_end; unsigned int i; unsigned int codecs_assigned; struct mgcp_rtp_codec *codec_convertible = NULL; rtp_end = &conn->end; /* Use the codec information from the source and try to find the equivalent of it on the destination side. In * the first run we will look for an exact match. */ codec_convertible = mgcp_codec_find_same(conn, codec); if (codec_convertible) return codec_convertible; /* In case we weren't able to find an exact match, we will try to find a match that is the same codec, but the * payload format may be different. This alternative will require a frame format conversion (i.e. AMR bwe->oe) */ codecs_assigned = rtp_end->codecs_assigned; OSMO_ASSERT(codecs_assigned <= MGCP_MAX_CODECS); for (i = 0; i < codecs_assigned; i++) { if (codecs_convertible(codec, &rtp_end->codecs[i])) { codec_convertible = &rtp_end->codecs[i]; break; } } return codec_convertible; } /*! Decide for one suitable codec on both of the given connections. In case a destination connection is not available, * a tentative decision is made. * \param[inout] conn_src related rtp-connection. * \param[inout] conn_dst related destination rtp-connection (NULL if not present). * \returns 0 on success, -EINVAL on failure. */ int mgcp_codec_decide(struct mgcp_conn_rtp *conn_src, struct mgcp_conn_rtp *conn_dst) { unsigned int i; /* In case no destination connection is available (yet), or in case the destination connection exists but has * no codecs assigned, we are forced to make a simple tentative decision: * We just use the first codec of the source connection (conn_src) */ OSMO_ASSERT(conn_src->end.codecs_assigned <= MGCP_MAX_CODECS); if (!conn_dst || conn_dst->end.codecs_assigned == 0) { if (conn_src->end.codecs_assigned >= 1) { conn_src->end.codec = &conn_src->end.codecs[0]; return 0; } else return -EINVAL; } /* Compare all codecs of the source connection (conn_src) to the codecs of the destination connection (conn_dst). In case * of a match set this codec on both connections. This would be an ideal selection since no codec conversion would be * required. */ for (i = 0; i < conn_src->end.codecs_assigned; i++) { struct mgcp_rtp_codec *codec_conn_src = &conn_src->end.codecs[i]; struct mgcp_rtp_codec *codec_conn_dst = mgcp_codec_find_same(conn_dst, codec_conn_src); if (codec_conn_dst) { /* We found the a codec that is exactly the same (same codec, same payload format etc.) on both * sides. We now set this codec on both connections. */ conn_dst->end.codec = codec_conn_dst; conn_src->end.codec = codec_conn_src; return 0; } } /* In case we could not find a codec that is exactly the same, let's at least try to find a codec that we are able * to convert. */ for (i = 0; i < conn_src->end.codecs_assigned; i++) { struct mgcp_rtp_codec *codec_conn_src = &conn_src->end.codecs[i]; struct mgcp_rtp_codec *codec_conn_dst = codec_find_convertible(conn_dst, codec_conn_src); if (codec_conn_dst) { /* We found the a codec that we can convert to. Set each side to its codec. */ conn_dst->end.codec = codec_conn_dst; conn_src->end.codec = codec_conn_src; return 0; } } if (conn_dst->end.codecs_assigned) conn_dst->end.codec = &conn_dst->end.codecs[0]; else return -EINVAL; if (conn_src->end.codecs_assigned) conn_src->end.codec = &conn_src->end.codecs[0]; else return -EINVAL; return 0; } /* Check if the codec has a specific AMR mode (octet-aligned or bandwith-efficient) set. */ bool mgcp_codec_amr_align_mode_is_indicated(const struct mgcp_rtp_codec *codec) { if (codec->param_present == false) return false; if (!codec->param.amr_octet_aligned_present) return false; if (strcmp(codec->subtype_name, "AMR") != 0) return false; return true; } /* Find the payload type number configured for a specific codec by SDP. * For example, IuUP gets assigned a payload type number, and the endpoint needs to translate that to the number * assigned to "AMR" on the other conn (by a=rtpmap:N). * \param conn The side of an endpoint to get the payload type number for (to translate the payload type number to). * \param subtype_name SDP codec name without parameters (e.g. "AMR"). * \param match_nr Index for the match found, first being match_nr == 0. Iterate all matches by calling multiple times * with incrementing match_nr. * \return codec definition for that conn matching the subtype_name, or NULL if no such match_nr is found. */ const struct mgcp_rtp_codec *mgcp_codec_pt_find_by_subtype_name(struct mgcp_conn_rtp *conn, const char *subtype_name, unsigned int match_nr) { int i; for (i = 0; i < conn->end.codecs_assigned; i++) { if (!strcmp(conn->end.codecs[i].subtype_name, subtype_name)) { if (match_nr) { match_nr--; continue; } return &conn->end.codecs[i]; } } return NULL; } /*! Lookup a codec that is assigned to a connection by its payload type number. * \param[in] conn related rtp-connection. * \param[in] payload_type number of the codec to look up. * \returns pointer to codec struct on success, NULL on failure. */ struct mgcp_rtp_codec *mgcp_codec_from_pt(struct mgcp_conn_rtp *conn, int payload_type) { struct mgcp_rtp_end *rtp_end = &conn->end; unsigned int codecs_assigned = rtp_end->codecs_assigned; struct mgcp_rtp_codec *codec = NULL; size_t i; OSMO_ASSERT(codecs_assigned <= MGCP_MAX_CODECS); for (i = 0; i < codecs_assigned; i++) { if (payload_type == rtp_end->codecs[i].payload_type) { codec = &rtp_end->codecs[i]; break; } } return codec; }