forked from osmocom/wireshark
![Michal Labedzki](/assets/img/avatar_default.png)
The Stream is defined here as media stream that beginning on AVDTP Start (ResponseAccept). Also fix recognizing Channel streams by AVDTP according to the specification that says: 1. First channel is always Signaling. 2. Second may be Media. 3. Third may be Reporting. 4. Fourth may be Recovery. First and second will be supported right now. Change-Id: Id6d4dae6be1b9df68382288c2d520b7ed3661237 Reviewed-on: https://code.wireshark.org/review/1053 Reviewed-by: Michal Labedzki <michal.labedzki@tieto.com>
216 lines
9.7 KiB
C
216 lines
9.7 KiB
C
/* packet-rtp.h
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*
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* Routines for RTP dissection
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* RTP = Real time Transport Protocol
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*
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* Copyright 2000, Philips Electronics N.V.
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* Written by Andreas Sikkema <andreas.sikkema@philips.com>
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*
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* Wireshark - Network traffic analyzer
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* By Gerald Combs <gerald@wireshark.org>
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* Copyright 1998 Gerald Combs
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "epan/packet.h"
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#include "ws_symbol_export.h"
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#include "packet-btavdtp.h"
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struct _rtp_info {
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unsigned int info_version;
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gboolean info_padding_set;
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gboolean info_marker_set;
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gboolean info_is_video;
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unsigned int info_payload_type;
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unsigned int info_padding_count;
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guint16 info_seq_num;
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guint32 info_timestamp;
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guint32 info_sync_src;
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guint info_data_len; /* length of raw rtp data as reported */
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gboolean info_all_data_present; /* FALSE if data is cut off */
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guint info_payload_offset; /* start of payload relative to info_data */
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guint info_payload_len; /* length of payload (incl padding) */
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gboolean info_is_srtp;
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guint32 info_setup_frame_num; /* the frame num of the packet that set this RTP connection */
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const guint8* info_data; /* pointer to raw rtp data */
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const gchar *info_payload_type_str;
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gint info_payload_rate;
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/*
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* info_data: pointer to raw rtp data = header + payload incl. padding.
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* That should be safe because the "epan_dissect_t" constructed for the packet
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* has not yet been freed when the taps are called.
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* (destroying the "epan_dissect_t" will end up freeing all the tvbuffs
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* and hence invalidating pointers to their data).
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* See "add_packet_to_packet_list()" for details.
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*/
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};
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/* definitions for SRTP dissection */
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/* Encryption algorithms */
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#define SRTP_ENC_ALG_NOT_SET 0 /* Data not available/empty record */
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#define SRTP_ENC_ALG_NULL 1 /* non-encrypted SRTP payload - may still be authenticated */
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#define SRTP_ENC_ALG_AES_CM 2 /* SRTP default algorithm */
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#define SRTP_ENC_ALG_AES_F8 3
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/* Authentication algorithms */
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#define SRTP_AUTH_ALG_NONE 0 /* no auth tag in SRTP/RTP payload */
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#define SRTP_AUTH_ALG_HMAC_SHA1 1 /* SRTP default algorithm */
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#if 0 /* these are only needed once the dissector include the crypto functions to decrypt and/or authenticate */
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struct srtp_key_info
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{
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guint8 *master_key; /* pointer to an se_alloc'ed master key */
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guint8 *master_salt; /* pointer to an se_alloc'ed salt for this master key - NULL if no salt */
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guint8 key_generation_rate; /* encoded as the power of 2, 0..24, or 255 (=zero rate) */
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/* Either the MKI value is used (in which case from=to=0), or the <from,to> values are used (and MKI=0) */
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guint32 from_roc; /* 32 MSBs of a 48 bit value - frame from which this key is valid (roll-over counter part) */
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guint16 from_seq; /* 16 LSBs of a 48 bit value - frame from which this key is valid (sequence number part) */
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guint32 to_roc; /* 32 MSBs of a 48 bit value - frame to which this key is valid (roll-over counter part) */
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guint16 to_seq; /* 16 LSBs of a 48 bit value - frame to which this key is valid (sequence number part) */
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guint32 mki; /* the MKI value associated with this key */
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};
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#endif
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struct srtp_info
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{
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guint encryption_algorithm; /* at present only NULL vs non-NULL matter */
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guint auth_algorithm; /* at present only NULL vs non-NULL matter */
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guint mki_len; /* number of octets used for the MKI in the RTP payload */
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guint auth_tag_len; /* number of octets used for the Auth Tag in the RTP payload */
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#if 0 /* these are only needed once the dissector include the crypto functions to decrypt and/or authenticate */
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struct srtp_key_info **master_keys; /* an array of pointers to master keys and their info, the array and each key struct being se_alloc'ed */
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void *enc_alg_info, /* algorithm-dependent info struct - may be void for default alg with default params */
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void *auth_alg_info /* algorithm-dependent info struct - void for default alg with default params */
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#endif
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};
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/* an opaque object holding the hash table - use accessor functions to create/destroy/find */
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typedef struct _rtp_dyn_payload_t rtp_dyn_payload_t;
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/* RTP dynamic payload handling - use the following to create, insert, lookup, and free the
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dynamic payload information. Internally, RTP creates the GHashTable with a wmem file scope
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and increments the ref_count when it saves the info to conversations later. The calling
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dissector (SDP, H.245, etc.) uses these functions as an interface. If the calling dissector
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is done with the rtp_dyn_payload_t* for good, it should call rtp_dyn_payload_free() which
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will decrement the ref_count and free's it if the ref_count is 0. In the worst case, it
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will get free'd when the wmem file scope is over.
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This was changed because there were too many bugs with SDP's handling of memory ownership
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of the GHashTable, with RTP freeing things SDP didn't think were free'ed. And also because
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the GHashTables never got free'd in many cases by several dissectors.
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*/
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/* creates a new hashtable and sets ref_count to 1, returning the newly created object */
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WS_DLL_PUBLIC
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rtp_dyn_payload_t* rtp_dyn_payload_new(void);
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/* Inserts the given payload type key, for the encoding name and sample rate, into the hash table.
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This makes copies of the encoding name, scoped to the life of the capture file or sooner if
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rtp_dyn_payload_free is called. */
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WS_DLL_PUBLIC
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void rtp_dyn_payload_insert(rtp_dyn_payload_t *rtp_dyn_payload,
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const guint pt,
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const gchar* encoding_name,
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const int sample_rate);
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/* Replaces the given payload type key in the hash table, with the encoding name and sample rate.
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This makes copies of the encoding name, scoped to the life of the capture file or sooner if
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rtp_dyn_payload_free is called. The replaced encoding name is free'd immediately. */
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WS_DLL_PUBLIC
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void rtp_dyn_payload_replace(rtp_dyn_payload_t *rtp_dyn_payload,
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const guint pt,
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const gchar* encoding_name,
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const int sample_rate);
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/* removes the given payload type */
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WS_DLL_PUBLIC
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gboolean rtp_dyn_payload_remove(rtp_dyn_payload_t *rtp_dyn_payload, const guint pt);
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/* retrieves the encoding name for the given payload type; the string returned is only valid
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until the entry is replaced, removed, or the hash table is destroyed, so duplicate it if
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you need it long. */
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WS_DLL_PUBLIC
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const gchar* rtp_dyn_payload_get_name(rtp_dyn_payload_t *rtp_dyn_payload, const guint pt);
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/* retrieves the encoding name and sample rate for the given payload type, returning TRUE if
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successful, else FALSE. The encoding string pointed to is only valid until the entry is
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replaced, removed, or the hash table is destroyed, so duplicate it if you need it long. */
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WS_DLL_PUBLIC
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gboolean rtp_dyn_payload_get_full(rtp_dyn_payload_t *rtp_dyn_payload, const guint pt,
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const gchar **encoding_name, int *sample_rate);
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/* Free's and destroys the dyn_payload hash table; internally this decrements the ref_count
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and only free's it if the ref_count == 0. */
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WS_DLL_PUBLIC
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void rtp_dyn_payload_free(rtp_dyn_payload_t *rtp_dyn_payload);
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#ifdef DEBUG_CONVERSATION
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/* used for printing out debugging info, if DEBUG_CONVERSATION is defined */
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void rtp_dump_dyn_payload(rtp_dyn_payload_t *rtp_dyn_payload);
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#endif
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/* Info to save in RTP conversation / packet-info */
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#define MAX_RTP_SETUP_METHOD_SIZE 7
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struct _rtp_conversation_info
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{
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gchar method[MAX_RTP_SETUP_METHOD_SIZE + 1];
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guint32 frame_number; /* the frame where this conversation is started */
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gboolean is_video;
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rtp_dyn_payload_t *rtp_dyn_payload; /* the dynamic RTP payload info - see comments above */
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guint32 extended_seqno; /* the sequence number, extended to a 32-bit
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* int to guarantee it increasing monotonically
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*/
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struct _rtp_private_conv_info *rtp_conv_info; /* conversation info private
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* to the rtp dissector
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*/
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struct srtp_info *srtp_info; /* SRTP context */
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bta2dp_codec_info_t *bta2dp_info;
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btvdp_codec_info_t *btvdp_info;
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};
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/* Add an RTP conversation with the given details */
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WS_DLL_PUBLIC
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void rtp_add_address(packet_info *pinfo,
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address *addr, int port,
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int other_port,
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const gchar *setup_method,
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guint32 setup_frame_number,
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gboolean is_video,
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rtp_dyn_payload_t *rtp_dyn_payload);
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/* Add an SRTP conversation with the given details */
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WS_DLL_PUBLIC
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void srtp_add_address(packet_info *pinfo,
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address *addr, int port,
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int other_port,
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const gchar *setup_method,
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guint32 setup_frame_number,
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gboolean is_video,
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rtp_dyn_payload_t *rtp_dyn_payload,
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struct srtp_info *srtp_info);
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/* Add an Bluetooth conversation with the given details */
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void
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bluetooth_add_address(packet_info *pinfo, address *addr, guint32 stream_number,
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const gchar *setup_method, guint32 setup_frame_number,
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gboolean is_video, void *data);
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