Add current version of libSRTP from CVS.

libsrtp/srtp/srtp.c

Index: libsrtp/srtp/srtp.c
===================================================================
--- libsrtp/srtp/srtp.c	(revision 0)
+++ libsrtp/srtp/srtp.c	(revision 0)
@@ -0,0 +1,2163 @@
+/*
+ * srtp.c
+ *
+ * the secure real-time transport protocol
+ *
+ * David A. McGrew
+ * Cisco Systems, Inc.
+ */
+/*
+ *	
+ * Copyright (c) 2001-2006, Cisco Systems, Inc.
+ * All rights reserved.
+ * 
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 
+ *   Redistributions of source code must retain the above copyright
+ *   notice, this list of conditions and the following disclaimer.
+ * 
+ *   Redistributions in binary form must reproduce the above
+ *   copyright notice, this list of conditions and the following
+ *   disclaimer in the documentation and/or other materials provided
+ *   with the distribution.
+ * 
+ *   Neither the name of the Cisco Systems, Inc. nor the names of its
+ *   contributors may be used to endorse or promote products derived
+ *   from this software without specific prior written permission.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ * FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
+ * COPYRIGHT HOLDERS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
+ * INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+ * STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
+ * OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+
+
+#include "srtp.h"
+#include "ekt.h"             /* for SRTP Encrypted Key Transport */
+#include "alloc.h"           /* for crypto_alloc()          */
+
+#ifndef SRTP_KERNEL
+# include <limits.h>
+# ifdef HAVE_NETINET_IN_H
+#  include <netinet/in.h>
+# elif defined(HAVE_WINSOCK2_H)
+#  include <winsock2.h>
+# endif
+#endif /* ! SRTP_KERNEL */
+
+
+/* the debug module for srtp */
+
+debug_module_t mod_srtp = {
+  0,                  /* debugging is off by default */
+  "srtp"              /* printable name for module   */
+};
+
+#define octets_in_rtp_header   12
+#define uint32s_in_rtp_header  3
+#define octets_in_rtcp_header  8
+#define uint32s_in_rtcp_header 2
+
+
+err_status_t
+srtp_stream_alloc(srtp_stream_ctx_t **str_ptr,
+		  const srtp_policy_t *p) {
+  srtp_stream_ctx_t *str;
+  err_status_t stat;
+
+  /*
+   * This function allocates the stream context, rtp and rtcp ciphers
+   * and auth functions, and key limit structure.  If there is a
+   * failure during allocation, we free all previously allocated
+   * memory and return a failure code.  The code could probably 
+   * be improved, but it works and should be clear.
+   */
+
+  /* allocate srtp stream and set str_ptr */
+  str = (srtp_stream_ctx_t *) crypto_alloc(sizeof(srtp_stream_ctx_t));
+  if (str == NULL)
+    return err_status_alloc_fail;
+  *str_ptr = str;  
+  
+  /* allocate cipher */
+  stat = crypto_kernel_alloc_cipher(p->rtp.cipher_type, 
+				    &str->rtp_cipher, 
+				    p->rtp.cipher_key_len); 
+  if (stat) {
+    crypto_free(str);
+    return stat;
+  }
+
+  /* allocate auth function */
+  stat = crypto_kernel_alloc_auth(p->rtp.auth_type, 
+				  &str->rtp_auth,
+				  p->rtp.auth_key_len, 
+				  p->rtp.auth_tag_len); 
+  if (stat) {
+    cipher_dealloc(str->rtp_cipher);
+    crypto_free(str);
+    return stat;
+  }
+  
+  /* allocate key limit structure */
+  str->limit = (key_limit_ctx_t*) crypto_alloc(sizeof(key_limit_ctx_t));
+  if (str->limit == NULL) {
+    auth_dealloc(str->rtp_auth);
+    cipher_dealloc(str->rtp_cipher);
+    crypto_free(str); 
+    return err_status_alloc_fail;
+  }
+
+  /*
+   * ...and now the RTCP-specific initialization - first, allocate
+   * the cipher 
+   */
+  stat = crypto_kernel_alloc_cipher(p->rtcp.cipher_type, 
+				    &str->rtcp_cipher, 
+				    p->rtcp.cipher_key_len); 
+  if (stat) {
+    auth_dealloc(str->rtp_auth);
+    cipher_dealloc(str->rtp_cipher);
+    crypto_free(str->limit);
+    crypto_free(str);
+    return stat;
+  }
+
+  /* allocate auth function */
+  stat = crypto_kernel_alloc_auth(p->rtcp.auth_type, 
+				  &str->rtcp_auth,
+				  p->rtcp.auth_key_len, 
+				  p->rtcp.auth_tag_len); 
+  if (stat) {
+    cipher_dealloc(str->rtcp_cipher);
+    auth_dealloc(str->rtp_auth);
+    cipher_dealloc(str->rtp_cipher);
+    crypto_free(str->limit);
+    crypto_free(str);
+   return stat;
+  }  
+
+  /* allocate ekt data associated with stream */
+  stat = ekt_alloc(&str->ekt, p->ekt);
+  if (stat) {
+    auth_dealloc(str->rtcp_auth);
+    cipher_dealloc(str->rtcp_cipher);
+    auth_dealloc(str->rtp_auth);
+    cipher_dealloc(str->rtp_cipher);
+    crypto_free(str->limit);
+    crypto_free(str);
+   return stat;    
+  }
+
+  return err_status_ok;
+}
+
+err_status_t
+srtp_stream_dealloc(srtp_t session, srtp_stream_ctx_t *stream) { 
+  err_status_t status;
+  
+  /*
+   * we use a conservative deallocation strategy - if any deallocation
+   * fails, then we report that fact without trying to deallocate
+   * anything else
+   */
+
+  /* deallocate cipher, if it is not the same as that in template */
+  if (session->stream_template
+      && stream->rtp_cipher == session->stream_template->rtp_cipher) {
+    /* do nothing */
+  } else {
+    status = cipher_dealloc(stream->rtp_cipher); 
+    if (status) 
+      return status;
+  }
+
+  /* deallocate auth function, if it is not the same as that in template */
+  if (session->stream_template
+      && stream->rtp_auth == session->stream_template->rtp_auth) {
+    /* do nothing */
+  } else {
+    status = auth_dealloc(stream->rtp_auth);
+    if (status)
+      return status;
+  }
+
+  /* deallocate key usage limit, if it is not the same as that in template */
+  if (session->stream_template
+      && stream->limit == session->stream_template->limit) {
+    /* do nothing */
+  } else {
+    crypto_free(stream->limit);
+  }   
+
+  /* 
+   * deallocate rtcp cipher, if it is not the same as that in
+   * template 
+   */
+  if (session->stream_template
+      && stream->rtcp_cipher == session->stream_template->rtcp_cipher) {
+    /* do nothing */
+  } else {
+    status = cipher_dealloc(stream->rtcp_cipher); 
+    if (status) 
+      return status;
+  }
+
+  /*
+   * deallocate rtcp auth function, if it is not the same as that in
+   * template 
+   */
+  if (session->stream_template
+      && stream->rtcp_auth == session->stream_template->rtcp_auth) {
+    /* do nothing */
+  } else {
+    status = auth_dealloc(stream->rtcp_auth);
+    if (status)
+      return status;
+  }
+
+  status = rdbx_dealloc(&stream->rtp_rdbx);
+  if (status)
+    return status;
+
+  /* DAM - need to deallocate EKT here */
+  
+  /* deallocate srtp stream context */
+  crypto_free(stream);
+
+  return err_status_ok;
+}
+
+
+/*
+ * srtp_stream_clone(stream_template, new) allocates a new stream and
+ * initializes it using the cipher and auth of the stream_template
+ * 
+ * the only unique data in a cloned stream is the replay database and
+ * the SSRC
+ */
+
+err_status_t
+srtp_stream_clone(const srtp_stream_ctx_t *stream_template, 
+		  uint32_t ssrc, 
+		  srtp_stream_ctx_t **str_ptr) {
+  err_status_t status;
+  srtp_stream_ctx_t *str;
+
+  debug_print(mod_srtp, "cloning stream (SSRC: 0x%08x)", ssrc);
+
+  /* allocate srtp stream and set str_ptr */
+  str = (srtp_stream_ctx_t *) crypto_alloc(sizeof(srtp_stream_ctx_t));
+  if (str == NULL)
+    return err_status_alloc_fail;
+  *str_ptr = str;  
+
+  /* set cipher and auth pointers to those of the template */
+  str->rtp_cipher  = stream_template->rtp_cipher;
+  str->rtp_auth    = stream_template->rtp_auth;
+  str->rtcp_cipher = stream_template->rtcp_cipher;
+  str->rtcp_auth   = stream_template->rtcp_auth;
+
+  /* set key limit to point to that of the template */
+  status = key_limit_clone(stream_template->limit, &str->limit);
+  if (status) 
+    return status;
+
+  /* initialize replay databases */
+  status = rdbx_init(&str->rtp_rdbx,
+		     rdbx_get_window_size(&stream_template->rtp_rdbx));
+  if (status)
+    return status;
+  rdb_init(&str->rtcp_rdb);
+  str->allow_repeat_tx = stream_template->allow_repeat_tx;
+  
+  /* set ssrc to that provided */
+  str->ssrc = ssrc;
+
+  /* set direction and security services */
+  str->direction     = stream_template->direction;
+  str->rtp_services  = stream_template->rtp_services;
+  str->rtcp_services = stream_template->rtcp_services;
+
+  /* set pointer to EKT data associated with stream */
+  str->ekt = stream_template->ekt;
+
+  /* defensive coding */
+  str->next = NULL;
+
+  return err_status_ok;
+}
+
+
+/*
+ * key derivation functions, internal to libSRTP
+ *
+ * srtp_kdf_t is a key derivation context
+ *
+ * srtp_kdf_init(&kdf, cipher_id, k, keylen) initializes kdf to use cipher
+ * described by cipher_id, with the master key k with length in octets keylen.
+ * 
+ * srtp_kdf_generate(&kdf, l, kl, keylen) derives the key
+ * corresponding to label l and puts it into kl; the length
+ * of the key in octets is provided as keylen.  this function
+ * should be called once for each subkey that is derived.
+ *
+ * srtp_kdf_clear(&kdf) zeroizes and deallocates the kdf state
+ */
+
+typedef enum {
+  label_rtp_encryption  = 0x00,
+  label_rtp_msg_auth    = 0x01,
+  label_rtp_salt        = 0x02,
+  label_rtcp_encryption = 0x03,
+  label_rtcp_msg_auth   = 0x04,
+  label_rtcp_salt       = 0x05
+} srtp_prf_label;
+
+
+/*
+ * srtp_kdf_t represents a key derivation function.  The SRTP
+ * default KDF is the only one implemented at present.
+ */
+
+typedef struct { 
+  cipher_t *cipher;    /* cipher used for key derivation  */  
+} srtp_kdf_t;
+
+err_status_t
+srtp_kdf_init(srtp_kdf_t *kdf, cipher_type_id_t cipher_id, const uint8_t *key, int length) {
+
+  err_status_t stat;
+  stat = crypto_kernel_alloc_cipher(cipher_id, &kdf->cipher, length);
+  if (stat)
+    return stat;
+
+  stat = cipher_init(kdf->cipher, key, direction_encrypt);
+  if (stat) {
+    cipher_dealloc(kdf->cipher);
+    return stat;
+  }
+
+  return err_status_ok;
+}
+
+err_status_t
+srtp_kdf_generate(srtp_kdf_t *kdf, srtp_prf_label label,
+		  uint8_t *key, unsigned length) {
+
+  v128_t nonce;
+  err_status_t status;
+  
+  /* set eigth octet of nonce to <label>, set the rest of it to zero */
+  v128_set_to_zero(&nonce);
+  nonce.v8[7] = label;
+ 
+  status = cipher_set_iv(kdf->cipher, &nonce);
+  if (status)
+    return status;
+  
+  /* generate keystream output */
+  octet_string_set_to_zero(key, length);
+  status = cipher_encrypt(kdf->cipher, key, &length);
+  if (status)
+    return status;
+
+  return err_status_ok;
+}
+
+err_status_t
+srtp_kdf_clear(srtp_kdf_t *kdf) {
+  err_status_t status;
+  status = cipher_dealloc(kdf->cipher);
+  if (status)
+    return status;
+  kdf->cipher = NULL;
+
+  return err_status_ok;  
+}
+
+/*
+ *  end of key derivation functions 
+ */
+
+#define MAX_SRTP_KEY_LEN 256
+
+
+/* Get the base key length corresponding to a given combined key+salt
+ * length for the given cipher.
+ * Assumption is that for AES-ICM a key length < 30 is Ismacryp using
+ * AES-128 and short salts; everything else uses a salt length of 14.
+ * TODO: key and salt lengths should be separate fields in the policy.  */
+inline int base_key_length(const cipher_type_t *cipher, int key_length)
+{
+  if (cipher->id != AES_ICM)
+    return key_length;
+  else if (key_length > 16 && key_length < 30)
+    return 16;
+  return key_length - 14;
+}
+
+err_status_t
+srtp_stream_init_keys(srtp_stream_ctx_t *srtp, const void *key) {
+  err_status_t stat;
+  srtp_kdf_t kdf;
+  uint8_t tmp_key[MAX_SRTP_KEY_LEN];
+  int kdf_keylen = 30, rtp_keylen, rtcp_keylen;
+  int rtp_base_key_len, rtp_salt_len;
+  int rtcp_base_key_len, rtcp_salt_len;
+
+  /* If RTP or RTCP have a key length > AES-128, assume matching kdf. */
+  /* TODO: kdf algorithm, master key length, and master salt length should
+   * be part of srtp_policy_t. */
+  rtp_keylen = cipher_get_key_length(srtp->rtp_cipher);
+  if (rtp_keylen > kdf_keylen)
+    kdf_keylen = rtp_keylen;
+
+  rtcp_keylen = cipher_get_key_length(srtp->rtcp_cipher);
+  if (rtcp_keylen > kdf_keylen)
+    kdf_keylen = rtcp_keylen;
+
+  /* initialize KDF state     */
+  stat = srtp_kdf_init(&kdf, AES_ICM, (const uint8_t *)key, kdf_keylen);
+  if (stat) {
+    return err_status_init_fail;
+  }
+
+  rtp_base_key_len = base_key_length(srtp->rtp_cipher->type, rtp_keylen);
+  rtp_salt_len = rtp_keylen - rtp_base_key_len;
+  
+  /* generate encryption key  */
+  stat = srtp_kdf_generate(&kdf, label_rtp_encryption, 
+			   tmp_key, rtp_base_key_len);
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+
+  /* 
+   * if the cipher in the srtp context uses a salt, then we need
+   * to generate the salt value
+   */
+  if (rtp_salt_len > 0) {
+    debug_print(mod_srtp, "found rtp_salt_len > 0, generating salt", NULL);
+
+    /* generate encryption salt, put after encryption key */
+    stat = srtp_kdf_generate(&kdf, label_rtp_salt, 
+			     tmp_key + rtp_base_key_len, rtp_salt_len);
+    if (stat) {
+      /* zeroize temp buffer */
+      octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+      return err_status_init_fail;
+    }
+  }
+  debug_print(mod_srtp, "cipher key: %s", 
+	      octet_string_hex_string(tmp_key, rtp_keylen));
+
+  /* initialize cipher */
+  stat = cipher_init(srtp->rtp_cipher, tmp_key, direction_any);
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+
+  /* generate authentication key */
+  stat = srtp_kdf_generate(&kdf, label_rtp_msg_auth,
+			   tmp_key, auth_get_key_length(srtp->rtp_auth));
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+  debug_print(mod_srtp, "auth key:   %s",
+	      octet_string_hex_string(tmp_key, 
+				      auth_get_key_length(srtp->rtp_auth))); 
+
+  /* initialize auth function */
+  stat = auth_init(srtp->rtp_auth, tmp_key);
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+
+  /*
+   * ...now initialize SRTCP keys
+   */
+
+  rtcp_base_key_len = base_key_length(srtp->rtcp_cipher->type, rtcp_keylen);
+  rtcp_salt_len = rtcp_keylen - rtcp_base_key_len;
+  
+  /* generate encryption key  */
+  stat = srtp_kdf_generate(&kdf, label_rtcp_encryption, 
+			   tmp_key, rtcp_base_key_len);
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+
+  /* 
+   * if the cipher in the srtp context uses a salt, then we need
+   * to generate the salt value
+   */
+  if (rtcp_salt_len > 0) {
+    debug_print(mod_srtp, "found rtcp_salt_len > 0, generating rtcp salt",
+		NULL);
+
+    /* generate encryption salt, put after encryption key */
+    stat = srtp_kdf_generate(&kdf, label_rtcp_salt, 
+			     tmp_key + rtcp_base_key_len, rtcp_salt_len);
+    if (stat) {
+      /* zeroize temp buffer */
+      octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+      return err_status_init_fail;
+    }
+  }
+  debug_print(mod_srtp, "rtcp cipher key: %s", 
+	      octet_string_hex_string(tmp_key, rtcp_keylen));  
+
+  /* initialize cipher */
+  stat = cipher_init(srtp->rtcp_cipher, tmp_key, direction_any);
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+
+  /* generate authentication key */
+  stat = srtp_kdf_generate(&kdf, label_rtcp_msg_auth,
+			   tmp_key, auth_get_key_length(srtp->rtcp_auth));
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+
+  debug_print(mod_srtp, "rtcp auth key:   %s",
+	      octet_string_hex_string(tmp_key, 
+		     auth_get_key_length(srtp->rtcp_auth))); 
+
+  /* initialize auth function */
+  stat = auth_init(srtp->rtcp_auth, tmp_key);
+  if (stat) {
+    /* zeroize temp buffer */
+    octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+    return err_status_init_fail;
+  }
+
+  /* clear memory then return */
+  stat = srtp_kdf_clear(&kdf);
+  octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);  
+  if (stat)
+    return err_status_init_fail;
+
+  return err_status_ok;
+}
+
+err_status_t
+srtp_stream_init(srtp_stream_ctx_t *srtp, 
+		  const srtp_policy_t *p) {
+  err_status_t err;
+
+   debug_print(mod_srtp, "initializing stream (SSRC: 0x%08x)", 
+	       p->ssrc.value);
+
+   /* initialize replay database */
+   /* window size MUST be at least 64.  MAY be larger.  Values more than
+    * 2^15 aren't meaningful due to how extended sequence numbers are
+    * calculated.   Let a window size of 0 imply the default value. */
+
+   if (p->window_size != 0 && (p->window_size < 64 || p->window_size >= 0x8000))
+     return err_status_bad_param;
+
+   if (p->window_size != 0)
+     err = rdbx_init(&srtp->rtp_rdbx, p->window_size);
+   else
+     err = rdbx_init(&srtp->rtp_rdbx, 128);
+   if (err) return err;
+
+   /* initialize key limit to maximum value */
+#ifdef NO_64BIT_MATH
+{
+   uint64_t temp;
+   temp = make64(UINT_MAX,UINT_MAX);
+   key_limit_set(srtp->limit, temp);
+}
+#else
+   key_limit_set(srtp->limit, 0xffffffffffffLL);
+#endif
+
+   /* set the SSRC value */
+   srtp->ssrc = htonl(p->ssrc.value);
+
+   /* set the security service flags */
+   srtp->rtp_services  = p->rtp.sec_serv;
+   srtp->rtcp_services = p->rtcp.sec_serv;
+
+   /*
+    * set direction to unknown - this flag gets checked in srtp_protect(),
+    * srtp_unprotect(), srtp_protect_rtcp(), and srtp_unprotect_rtcp(), and 
+    * gets set appropriately if it is set to unknown.
+    */
+   srtp->direction = dir_unknown;
+
+   /* initialize SRTCP replay database */
+   rdb_init(&srtp->rtcp_rdb);
+
+   /* initialize allow_repeat_tx */
+   /* guard against uninitialized memory: allow only 0 or 1 here */
+   if (p->allow_repeat_tx != 0 && p->allow_repeat_tx != 1) {
+     rdbx_dealloc(&srtp->rtp_rdbx);
+     return err_status_bad_param;
+   }
+   srtp->allow_repeat_tx = p->allow_repeat_tx;
+
+   /* DAM - no RTCP key limit at present */
+
+   /* initialize keys */
+   err = srtp_stream_init_keys(srtp, p->key);
+   if (err) {
+     rdbx_dealloc(&srtp->rtp_rdbx);
+     return err;
+   }
+
+   /* 
+    * if EKT is in use, then initialize the EKT data associated with
+    * the stream
+    */
+   err = ekt_stream_init_from_policy(srtp->ekt, p->ekt);
+   if (err) {
+     rdbx_dealloc(&srtp->rtp_rdbx);
+     return err;
+   }
+
+   return err_status_ok;  
+ }
+
+
+ /*
+  * srtp_event_reporter is an event handler function that merely
+  * reports the events that are reported by the callbacks
+  */
+
+ void
+ srtp_event_reporter(srtp_event_data_t *data) {
+
+   err_report(err_level_warning, "srtp: in stream 0x%x: ", 
+	      data->stream->ssrc);
+
+   switch(data->event) {
+   case event_ssrc_collision:
+     err_report(err_level_warning, "\tSSRC collision\n");
+     break;
+   case event_key_soft_limit:
+     err_report(err_level_warning, "\tkey usage soft limit reached\n");
+     break;
+   case event_key_hard_limit:
+     err_report(err_level_warning, "\tkey usage hard limit reached\n");
+     break;
+   case event_packet_index_limit:
+     err_report(err_level_warning, "\tpacket index limit reached\n");
+     break;
+   default:
+     err_report(err_level_warning, "\tunknown event reported to handler\n");
+   }
+ }
+
+ /*
+  * srtp_event_handler is a global variable holding a pointer to the
+  * event handler function; this function is called for any unexpected
+  * event that needs to be handled out of the SRTP data path.  see
+  * srtp_event_t in srtp.h for more info
+  *
+  * it is okay to set srtp_event_handler to NULL, but we set 
+  * it to the srtp_event_reporter.
+  */
+
+ static srtp_event_handler_func_t *srtp_event_handler = srtp_event_reporter;
+
+ err_status_t
+ srtp_install_event_handler(srtp_event_handler_func_t func) {
+
+   /* 
+    * note that we accept NULL arguments intentionally - calling this
+    * function with a NULL arguments removes an event handler that's
+    * been previously installed
+    */
+
+   /* set global event handling function */
+   srtp_event_handler = func;
+   return err_status_ok;
+ }
+
+ err_status_t
+ srtp_protect(srtp_ctx_t *ctx, void *rtp_hdr, int *pkt_octet_len) {
+   srtp_hdr_t *hdr = (srtp_hdr_t *)rtp_hdr;
+   uint32_t *enc_start;        /* pointer to start of encrypted portion  */
+   uint32_t *auth_start;       /* pointer to start of auth. portion      */
+   unsigned enc_octet_len = 0; /* number of octets in encrypted portion  */
+   xtd_seq_num_t est;          /* estimated xtd_seq_num_t of *hdr        */
+   int delta;                  /* delta of local pkt idx and that in hdr */
+   uint8_t *auth_tag = NULL;   /* location of auth_tag within packet     */
+   err_status_t status;   
+   int tag_len;
+   srtp_stream_ctx_t *stream;
+   int prefix_len;
+
+   debug_print(mod_srtp, "function srtp_protect", NULL);
+
+  /* we assume the hdr is 32-bit aligned to start */
+
+   /* check the packet length - it must at least contain a full header */
+   if (*pkt_octet_len < octets_in_rtp_header)
+     return err_status_bad_param;
+
+   /*
+    * look up ssrc in srtp_stream list, and process the packet with
+    * the appropriate stream.  if we haven't seen this stream before,
+    * there's a template key for this srtp_session, and the cipher
+    * supports key-sharing, then we assume that a new stream using
+    * that key has just started up
+    */
+   stream = srtp_get_stream(ctx, hdr->ssrc);
+   if (stream == NULL) {
+     if (ctx->stream_template != NULL) {
+       srtp_stream_ctx_t *new_stream;
+
+       /* allocate and initialize a new stream */
+       status = srtp_stream_clone(ctx->stream_template, 
+				  hdr->ssrc, &new_stream); 
+       if (status)
+	 return status;
+
+       /* add new stream to the head of the stream_list */
+       new_stream->next = ctx->stream_list;
+       ctx->stream_list = new_stream;
+
+       /* set direction to outbound */
+       new_stream->direction = dir_srtp_sender;
+
+       /* set stream (the pointer used in this function) */
+       stream = new_stream;
+     } else {
+       /* no template stream, so we return an error */
+       return err_status_no_ctx;
+     } 
+   }
+
+   /* 
+    * verify that stream is for sending traffic - this check will
+    * detect SSRC collisions, since a stream that appears in both
+    * srtp_protect() and srtp_unprotect() will fail this test in one of
+    * those functions.
+    */
+   if (stream->direction != dir_srtp_sender) {
+     if (stream->direction == dir_unknown) {
+       stream->direction = dir_srtp_sender;
+     } else {
+       srtp_handle_event(ctx, stream, event_ssrc_collision);
+     }
+   }
+
+  /* 
+   * update the key usage limit, and check it to make sure that we
+   * didn't just hit either the soft limit or the hard limit, and call
+   * the event handler if we hit either.
+   */
+  switch(key_limit_update(stream->limit)) {
+  case key_event_normal:
+    break;
+  case key_event_soft_limit: 
+    srtp_handle_event(ctx, stream, event_key_soft_limit);
+    break; 
+  case key_event_hard_limit:
+    srtp_handle_event(ctx, stream, event_key_hard_limit);
+	return err_status_key_expired;
+  default:
+    break;
+  }
+
+   /* get tag length from stream */
+   tag_len = auth_get_tag_length(stream->rtp_auth); 
+
+   /*
+    * find starting point for encryption and length of data to be
+    * encrypted - the encrypted portion starts after the rtp header
+    * extension, if present; otherwise, it starts after the last csrc,
+    * if any are present
+    *
+    * if we're not providing confidentiality, set enc_start to NULL
+    */
+   if (stream->rtp_services & sec_serv_conf) {
+     enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;  
+     if (hdr->x == 1) {
+       srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
+       enc_start += (ntohs(xtn_hdr->length) + 1);
+     }
+     enc_octet_len = (unsigned int)(*pkt_octet_len 
+				    - ((enc_start - (uint32_t *)hdr) << 2));
+   } else {
+     enc_start = NULL;
+   }
+
+   /* 
+    * if we're providing authentication, set the auth_start and auth_tag
+    * pointers to the proper locations; otherwise, set auth_start to NULL
+    * to indicate that no authentication is needed
+    */
+   if (stream->rtp_services & sec_serv_auth) {
+     auth_start = (uint32_t *)hdr;
+     auth_tag = (uint8_t *)hdr + *pkt_octet_len;
+   } else {
+     auth_start = NULL;
+     auth_tag = NULL;
+   }
+
+   /*
+    * estimate the packet index using the start of the replay window   
+    * and the sequence number from the header
+    */
+   delta = rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq));
+   status = rdbx_check(&stream->rtp_rdbx, delta);
+   if (status) {
+     if (status != err_status_replay_fail || !stream->allow_repeat_tx)
+       return status;  /* we've been asked to reuse an index */
+   }
+   else
+     rdbx_add_index(&stream->rtp_rdbx, delta);
+
+#ifdef NO_64BIT_MATH
+   debug_print2(mod_srtp, "estimated packet index: %08x%08x", 
+		high32(est),low32(est));
+#else
+   debug_print(mod_srtp, "estimated packet index: %016llx", est);
+#endif
+
+   /* 
+    * if we're using rindael counter mode, set nonce and seq 
+    */
+   if (stream->rtp_cipher->type->id == AES_ICM) {
+     v128_t iv;
+
+     iv.v32[0] = 0;
+     iv.v32[1] = hdr->ssrc;
+#ifdef NO_64BIT_MATH
+     iv.v64[1] = be64_to_cpu(make64((high32(est) << 16) | (low32(est) >> 16),
+								 low32(est) << 16));
+#else
+     iv.v64[1] = be64_to_cpu(est << 16);
+#endif
+     status = cipher_set_iv(stream->rtp_cipher, &iv);
+
+   } else {  
+     v128_t iv;
+
+     /* otherwise, set the index to est */  
+#ifdef NO_64BIT_MATH
+     iv.v32[0] = 0;
+     iv.v32[1] = 0;
+#else
+     iv.v64[0] = 0;
+#endif
+     iv.v64[1] = be64_to_cpu(est);
+     status = cipher_set_iv(stream->rtp_cipher, &iv);
+   }
+   if (status)
+     return err_status_cipher_fail;
+
+   /* shift est, put into network byte order */
+#ifdef NO_64BIT_MATH
+   est = be64_to_cpu(make64((high32(est) << 16) |
+						 (low32(est) >> 16),
+						 low32(est) << 16));
+#else
+   est = be64_to_cpu(est << 16);
+#endif
+   
+   /* 
+    * if we're authenticating using a universal hash, put the keystream
+    * prefix into the authentication tag
+    */
+   if (auth_start) {
+     
+    prefix_len = auth_get_prefix_length(stream->rtp_auth);    
+    if (prefix_len) {
+      status = cipher_output(stream->rtp_cipher, auth_tag, prefix_len);
+      if (status)
+	return err_status_cipher_fail;
+      debug_print(mod_srtp, "keystream prefix: %s", 
+		  octet_string_hex_string(auth_tag, prefix_len));
+    }
+  }
+
+  /* if we're encrypting, exor keystream into the message */
+  if (enc_start) {
+    status = cipher_encrypt(stream->rtp_cipher, 
+			    (uint8_t *)enc_start, &enc_octet_len);
+    if (status)
+      return err_status_cipher_fail;
+  }
+
+  /*
+   *  if we're authenticating, run authentication function and put result
+   *  into the auth_tag 
+   */
+  if (auth_start) {        
+
+    /* initialize auth func context */
+    status = auth_start(stream->rtp_auth);
+    if (status) return status;
+
+    /* run auth func over packet */
+    status = auth_update(stream->rtp_auth, 
+			 (uint8_t *)auth_start, *pkt_octet_len);
+    if (status) return status;
+    
+    /* run auth func over ROC, put result into auth_tag */
+    debug_print(mod_srtp, "estimated packet index: %016llx", est);
+    status = auth_compute(stream->rtp_auth, (uint8_t *)&est, 4, auth_tag); 
+    debug_print(mod_srtp, "srtp auth tag:    %s", 
+		octet_string_hex_string(auth_tag, tag_len));
+    if (status)
+      return err_status_auth_fail;   
+
+  }
+
+  if (auth_tag) {
+
+    /* increase the packet length by the length of the auth tag */
+    *pkt_octet_len += tag_len;
+  }
+
+  return err_status_ok;  
+}
+
+
+err_status_t
+srtp_unprotect(srtp_ctx_t *ctx, void *srtp_hdr, int *pkt_octet_len) {
+  srtp_hdr_t *hdr = (srtp_hdr_t *)srtp_hdr;
+  uint32_t *enc_start;      /* pointer to start of encrypted portion  */
+  uint32_t *auth_start;     /* pointer to start of auth. portion      */
+  unsigned enc_octet_len = 0;/* number of octets in encrypted portion */
+  uint8_t *auth_tag = NULL; /* location of auth_tag within packet     */
+  xtd_seq_num_t est;        /* estimated xtd_seq_num_t of *hdr        */
+  int delta;                /* delta of local pkt idx and that in hdr */
+  v128_t iv;
+  err_status_t status;
+  srtp_stream_ctx_t *stream;
+  uint8_t tmp_tag[SRTP_MAX_TAG_LEN];
+  int tag_len, prefix_len;
+
+  debug_print(mod_srtp, "function srtp_unprotect", NULL);
+
+  /* we assume the hdr is 32-bit aligned to start */
+
+  /* check the packet length - it must at least contain a full header */
+  if (*pkt_octet_len < octets_in_rtp_header)
+    return err_status_bad_param;
+
+  /*
+   * look up ssrc in srtp_stream list, and process the packet with 
+   * the appropriate stream.  if we haven't seen this stream before,
+   * there's only one key for this srtp_session, and the cipher
+   * supports key-sharing, then we assume that a new stream using
+   * that key has just started up
+   */
+  stream = srtp_get_stream(ctx, hdr->ssrc);
+  if (stream == NULL) {
+    if (ctx->stream_template != NULL) {
+      stream = ctx->stream_template;
+      debug_print(mod_srtp, "using provisional stream (SSRC: 0x%08x)",
+		  hdr->ssrc);
+      
+      /* 
+       * set estimated packet index to sequence number from header,
+       * and set delta equal to the same value
+       */
+#ifdef NO_64BIT_MATH
+      est = (xtd_seq_num_t) make64(0,ntohs(hdr->seq));
+      delta = low32(est);
+#else
+      est = (xtd_seq_num_t) ntohs(hdr->seq);
+      delta = (int)est;
+#endif
+    } else {
+      
+      /*
+       * no stream corresponding to SSRC found, and we don't do
+       * key-sharing, so return an error
+       */
+      return err_status_no_ctx;
+    }
+  } else {
+  
+    /* estimate packet index from seq. num. in header */
+    delta = rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq));
+    
+    /* check replay database */
+    status = rdbx_check(&stream->rtp_rdbx, delta);
+    if (status)
+      return status;
+  }
+
+#ifdef NO_64BIT_MATH
+  debug_print2(mod_srtp, "estimated u_packet index: %08x%08x", high32(est),low32(est));
+#else
+  debug_print(mod_srtp, "estimated u_packet index: %016llx", est);
+#endif
+
+  /* get tag length from stream */
+  tag_len = auth_get_tag_length(stream->rtp_auth); 
+
+  /* 
+   * set the cipher's IV properly, depending on whatever cipher we
+   * happen to be using
+   */
+  if (stream->rtp_cipher->type->id == AES_ICM) {
+
+    /* aes counter mode */
+    iv.v32[0] = 0;
+    iv.v32[1] = hdr->ssrc;  /* still in network order */
+#ifdef NO_64BIT_MATH
+    iv.v64[1] = be64_to_cpu(make64((high32(est) << 16) | (low32(est) >> 16),
+			         low32(est) << 16));
+#else
+    iv.v64[1] = be64_to_cpu(est << 16);
+#endif
+    status = cipher_set_iv(stream->rtp_cipher, &iv);
+  } else {  
+    
+    /* no particular format - set the iv to the pakcet index */  
+#ifdef NO_64BIT_MATH
+    iv.v32[0] = 0;
+    iv.v32[1] = 0;
+#else
+    iv.v64[0] = 0;
+#endif
+    iv.v64[1] = be64_to_cpu(est);
+    status = cipher_set_iv(stream->rtp_cipher, &iv);
+  }
+  if (status)
+    return err_status_cipher_fail;
+
+  /* shift est, put into network byte order */
+#ifdef NO_64BIT_MATH
+  est = be64_to_cpu(make64((high32(est) << 16) |
+					    (low32(est) >> 16),
+					    low32(est) << 16));
+#else
+  est = be64_to_cpu(est << 16);
+#endif
+
+  /*
+   * find starting point for decryption and length of data to be
+   * decrypted - the encrypted portion starts after the rtp header
+   * extension, if present; otherwise, it starts after the last csrc,
+   * if any are present
+   *
+   * if we're not providing confidentiality, set enc_start to NULL
+   */
+  if (stream->rtp_services & sec_serv_conf) {
+    enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;  
+    if (hdr->x == 1) {
+      srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
+      enc_start += (ntohs(xtn_hdr->length) + 1);
+    }  
+    enc_octet_len = (uint32_t)(*pkt_octet_len - tag_len 
+			       - ((enc_start - (uint32_t *)hdr) << 2));
+  } else {
+    enc_start = NULL;
+  }
+
+  /* 
+   * if we're providing authentication, set the auth_start and auth_tag
+   * pointers to the proper locations; otherwise, set auth_start to NULL
+   * to indicate that no authentication is needed
+   */
+  if (stream->rtp_services & sec_serv_auth) {
+    auth_start = (uint32_t *)hdr;
+    auth_tag = (uint8_t *)hdr + *pkt_octet_len - tag_len;
+  } else {
+    auth_start = NULL;
+    auth_tag = NULL;
+  } 
+
+  /*
+   * if we expect message authentication, run the authentication
+   * function and compare the result with the value of the auth_tag
+   */
+  if (auth_start) {        
+
+    /* 
+     * if we're using a universal hash, then we need to compute the
+     * keystream prefix for encrypting the universal hash output
+     *
+     * if the keystream prefix length is zero, then we know that
+     * the authenticator isn't using a universal hash function
+     */  
+    if (stream->rtp_auth->prefix_len != 0) {
+      
+      prefix_len = auth_get_prefix_length(stream->rtp_auth);    
+      status = cipher_output(stream->rtp_cipher, tmp_tag, prefix_len);
+      debug_print(mod_srtp, "keystream prefix: %s", 
+		  octet_string_hex_string(tmp_tag, prefix_len));
+      if (status)
+	return err_status_cipher_fail;
+    } 
+
+    /* initialize auth func context */
+    status = auth_start(stream->rtp_auth);
+    if (status) return status;
+ 
+    /* now compute auth function over packet */
+    status = auth_update(stream->rtp_auth, (uint8_t *)auth_start,  
+			 *pkt_octet_len - tag_len);
+
+    /* run auth func over ROC, then write tmp tag */
+    status = auth_compute(stream->rtp_auth, (uint8_t *)&est, 4, tmp_tag);  
+
+    debug_print(mod_srtp, "computed auth tag:    %s", 
+		octet_string_hex_string(tmp_tag, tag_len));
+    debug_print(mod_srtp, "packet auth tag:      %s", 
+		octet_string_hex_string(auth_tag, tag_len));
+    if (status)
+      return err_status_auth_fail;   
+
+    if (octet_string_is_eq(tmp_tag, auth_tag, tag_len))
+      return err_status_auth_fail;
+  }
+
+  /* 
+   * update the key usage limit, and check it to make sure that we
+   * didn't just hit either the soft limit or the hard limit, and call
+   * the event handler if we hit either.
+   */
+  switch(key_limit_update(stream->limit)) {
+  case key_event_normal:
+    break;
+  case key_event_soft_limit: 
+    srtp_handle_event(ctx, stream, event_key_soft_limit);
+    break; 
+  case key_event_hard_limit:
+    srtp_handle_event(ctx, stream, event_key_hard_limit);
+    return err_status_key_expired;
+  default:
+    break;
+  }
+
+  /* if we're decrypting, add keystream into ciphertext */
+  if (enc_start) {
+    status = cipher_decrypt(stream->rtp_cipher, 
+			    (uint8_t *)enc_start, &enc_octet_len);
+    if (status)
+      return err_status_cipher_fail;
+  }
+
+  /* 
+   * verify that stream is for received traffic - this check will
+   * detect SSRC collisions, since a stream that appears in both
+   * srtp_protect() and srtp_unprotect() will fail this test in one of
+   * those functions.
+   *
+   * we do this check *after* the authentication check, so that the
+   * latter check will catch any attempts to fool us into thinking
+   * that we've got a collision
+   */
+  if (stream->direction != dir_srtp_receiver) {
+    if (stream->direction == dir_unknown) {
+      stream->direction = dir_srtp_receiver;
+    } else {
+      srtp_handle_event(ctx, stream, event_ssrc_collision);
+    }
+  }
+
+  /* 
+   * if the stream is a 'provisional' one, in which the template context
+   * is used, then we need to allocate a new stream at this point, since
+   * the authentication passed
+   */
+  if (stream == ctx->stream_template) {  
+    srtp_stream_ctx_t *new_stream;
+
+    /* 
+     * allocate and initialize a new stream 
+     * 
+     * note that we indicate failure if we can't allocate the new
+     * stream, and some implementations will want to not return
+     * failure here
+     */
+    status = srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream); 
+    if (status)
+      return status;
+    
+    /* add new stream to the head of the stream_list */
+    new_stream->next = ctx->stream_list;
+    ctx->stream_list = new_stream;
+    
+    /* set stream (the pointer used in this function) */
+    stream = new_stream;
+  }
+  
+  /* 
+   * the message authentication function passed, so add the packet
+   * index into the replay database 
+   */
+  rdbx_add_index(&stream->rtp_rdbx, delta);
+
+  /* decrease the packet length by the length of the auth tag */
+  *pkt_octet_len -= tag_len;
+
+  return err_status_ok;  
+}
+
+err_status_t
+srtp_init() {
+  err_status_t status;
+
+  /* initialize crypto kernel */
+  status = crypto_kernel_init();
+  if (status) 
+    return status;
+
+  /* load srtp debug module into the kernel */
+  status = crypto_kernel_load_debug_module(&mod_srtp);
+  if (status)
+    return status;
+
+  return err_status_ok;
+}
+
+err_status_t
+srtp_shutdown() {
+  err_status_t status;
+
+  /* shut down crypto kernel */
+  status = crypto_kernel_shutdown();
+  if (status) 
+    return status;
+
+  /* shutting down crypto kernel frees the srtp debug module as well */
+
+  return err_status_ok;
+}
+
+
+/* 
+ * The following code is under consideration for removal.  See
+ * SRTP_MAX_TRAILER_LEN 
+ */
+#if 0
+
+/*
+ * srtp_get_trailer_length(&a) returns the number of octets that will
+ * be added to an RTP packet by the SRTP processing.  This value
+ * is constant for a given srtp_stream_t (i.e. between initializations).
+ */
+
+int
+srtp_get_trailer_length(const srtp_stream_t s) {
+  return auth_get_tag_length(s->rtp_auth);
+}
+
+#endif
+
+/*
+ * srtp_get_stream(ssrc) returns a pointer to the stream corresponding
+ * to ssrc, or NULL if no stream exists for that ssrc
+ *
+ * this is an internal function 
+ */
+
+srtp_stream_ctx_t *
+srtp_get_stream(srtp_t srtp, uint32_t ssrc) {
+  srtp_stream_ctx_t *stream;
+
+  /* walk down list until ssrc is found */
+  stream = srtp->stream_list;
+  while (stream != NULL) {
+    if (stream->ssrc == ssrc)
+      return stream;
+    stream = stream->next;
+  }
+  
+  /* we haven't found our ssrc, so return a null */
+  return NULL;
+}
+
+err_status_t
+srtp_dealloc(srtp_t session) {
+  srtp_stream_ctx_t *stream;
+  err_status_t status;
+
+  /*
+   * we take a conservative deallocation strategy - if we encounter an
+   * error deallocating a stream, then we stop trying to deallocate
+   * memory and just return an error
+   */
+
+  /* walk list of streams, deallocating as we go */
+  stream = session->stream_list;
+  while (stream != NULL) {
+    srtp_stream_t next = stream->next;
+    status = srtp_stream_dealloc(session, stream);
+    if (status)
+      return status;
+    stream = next;
+  }
+  
+  /* deallocate stream template, if there is one */
+  if (session->stream_template != NULL) {
+    status = auth_dealloc(session->stream_template->rtcp_auth); 
+    if (status) 
+      return status; 
+    status = cipher_dealloc(session->stream_template->rtcp_cipher); 
+    if (status) 
+      return status; 
+    crypto_free(session->stream_template->limit);
+    status = cipher_dealloc(session->stream_template->rtp_cipher); 
+    if (status) 
+      return status; 
+    status = auth_dealloc(session->stream_template->rtp_auth);
+    if (status)
+      return status;
+    status = rdbx_dealloc(&session->stream_template->rtp_rdbx);
+    if (status)
+      return status;
+    crypto_free(session->stream_template);
+  }
+
+  /* deallocate session context */
+  crypto_free(session);
+
+  return err_status_ok;
+}
+
+
+err_status_t
+srtp_add_stream(srtp_t session, 
+		const srtp_policy_t *policy)  {
+  err_status_t status;
+  srtp_stream_t tmp;
+
+  /* sanity check arguments */
+  if ((session == NULL) || (policy == NULL) || (policy->key == NULL))
+    return err_status_bad_param;
+
+  /* allocate stream  */
+  status = srtp_stream_alloc(&tmp, policy);
+  if (status) {
+    return status;
+  }
+  
+  /* initialize stream  */
+  status = srtp_stream_init(tmp, policy);
+  if (status) {
+    crypto_free(tmp);
+    return status;
+  }
+  
+  /* 
+   * set the head of the stream list or the template to point to the
+   * stream that we've just alloced and init'ed, depending on whether
+   * or not it has a wildcard SSRC value or not
+   *
+   * if the template stream has already been set, then the policy is
+   * inconsistent, so we return a bad_param error code
+   */
+  switch (policy->ssrc.type) {
+  case (ssrc_any_outbound):
+    if (session->stream_template) {
+      return err_status_bad_param;
+    }
+    session->stream_template = tmp;
+    session->stream_template->direction = dir_srtp_sender;
+    break;
+  case (ssrc_any_inbound):
+    if (session->stream_template) {
+      return err_status_bad_param;
+    }
+    session->stream_template = tmp;
+    session->stream_template->direction = dir_srtp_receiver;
+    break;
+  case (ssrc_specific):
+    tmp->next = session->stream_list;
+    session->stream_list = tmp;
+    break;
+  case (ssrc_undefined):
+  default:
+    crypto_free(tmp);
+    return err_status_bad_param;
+  }
+    
+  return err_status_ok;
+}
+
+
+err_status_t
+srtp_create(srtp_t *session,               /* handle for session     */ 
+	    const srtp_policy_t *policy) { /* SRTP policy (list)     */
+  err_status_t stat;
+  srtp_ctx_t *ctx;
+
+  /* sanity check arguments */
+  if (session == NULL)
+    return err_status_bad_param;
+
+  /* allocate srtp context and set ctx_ptr */
+  ctx = (srtp_ctx_t *) crypto_alloc(sizeof(srtp_ctx_t));
+  if (ctx == NULL)
+    return err_status_alloc_fail;
+  *session = ctx;
+
+  /* 
+   * loop over elements in the policy list, allocating and
+   * initializing a stream for each element
+   */
+  ctx->stream_template = NULL;
+  ctx->stream_list = NULL;
+  while (policy != NULL) {    
+
+    stat = srtp_add_stream(ctx, policy);
+    if (stat) {
+      /* clean up everything */
+      srtp_dealloc(*session);
+      return stat;
+    }    
+
+    /* set policy to next item in list  */
+    policy = policy->next;
+  }
+
+  return err_status_ok;
+}
+
+
+err_status_t
+srtp_remove_stream(srtp_t session, uint32_t ssrc) {
+  srtp_stream_ctx_t *stream, *last_stream;
+  err_status_t status;
+
+  /* sanity check arguments */
+  if (session == NULL)
+    return err_status_bad_param;
+  
+  /* find stream in list; complain if not found */
+  last_stream = stream = session->stream_list;
+  while ((stream != NULL) && (ssrc != stream->ssrc)) {
+    last_stream = stream;
+    stream = stream->next;
+  }
+  if (stream == NULL)
+    return err_status_no_ctx;
+
+  /* remove stream from the list */
+  if (last_stream == stream)
+    /* stream was first in list */
+    session->stream_list = stream->next;
+  else
+    last_stream->next = stream->next;
+
+  /* deallocate the stream */
+  status = srtp_stream_dealloc(session, stream);
+  if (status)
+    return status;
+
+  return err_status_ok;
+}
+
+
+/*
+ * the default policy - provides a convenient way for callers to use
+ * the default security policy
+ * 
+ * this policy is that defined in the current SRTP internet draft.
+ *
+ */
+
+/* 
+ * NOTE: cipher_key_len is really key len (128 bits) plus salt len
+ *  (112 bits)
+ */
+/* There are hard-coded 16's for base_key_len in the key generation code */
+
+void
+crypto_policy_set_rtp_default(crypto_policy_t *p) {
+
+  p->cipher_type     = AES_ICM;           
+  p->cipher_key_len  = 30;                /* default 128 bits per RFC 3711 */
+  p->auth_type       = HMAC_SHA1;             
+  p->auth_key_len    = 20;                /* default 160 bits per RFC 3711 */
+  p->auth_tag_len    = 10;                /* default 80 bits per RFC 3711 */
+  p->sec_serv        = sec_serv_conf_and_auth;
+  
+}
+
+void
+crypto_policy_set_rtcp_default(crypto_policy_t *p) {
+
+  p->cipher_type     = AES_ICM;           
+  p->cipher_key_len  = 30;                 /* default 128 bits per RFC 3711 */
+  p->auth_type       = HMAC_SHA1;             
+  p->auth_key_len    = 20;                 /* default 160 bits per RFC 3711 */
+  p->auth_tag_len    = 10;                 /* default 80 bits per RFC 3711 */
+  p->sec_serv        = sec_serv_conf_and_auth;
+  
+}
+
+void
+crypto_policy_set_aes_cm_128_hmac_sha1_32(crypto_policy_t *p) {
+
+  /*
+   * corresponds to RFC 4568
+   *
+   * note that this crypto policy is intended for SRTP, but not SRTCP
+   */
+
+  p->cipher_type     = AES_ICM;           
+  p->cipher_key_len  = 30;                /* 128 bit key, 112 bit salt */
+  p->auth_type       = HMAC_SHA1;             
+  p->auth_key_len    = 20;                /* 160 bit key               */
+  p->auth_tag_len    = 4;                 /* 32 bit tag                */
+  p->sec_serv        = sec_serv_conf_and_auth;
+  
+}
+
+
+void
+crypto_policy_set_aes_cm_128_null_auth(crypto_policy_t *p) {
+
+  /*
+   * corresponds to RFC 4568
+   *
+   * note that this crypto policy is intended for SRTP, but not SRTCP
+   */
+
+  p->cipher_type     = AES_ICM;           
+  p->cipher_key_len  = 30;                /* 128 bit key, 112 bit salt */
+  p->auth_type       = NULL_AUTH;             
+  p->auth_key_len    = 0; 
+  p->auth_tag_len    = 0; 
+  p->sec_serv        = sec_serv_conf;
+  
+}
+
+
+void
+crypto_policy_set_null_cipher_hmac_sha1_80(crypto_policy_t *p) {
+
+  /*
+   * corresponds to RFC 4568
+   */
+
+  p->cipher_type     = NULL_CIPHER;           
+  p->cipher_key_len  = 0;
+  p->auth_type       = HMAC_SHA1;             
+  p->auth_key_len    = 20; 
+  p->auth_tag_len    = 10; 
+  p->sec_serv        = sec_serv_auth;
+  
+}
+
+
+void
+crypto_policy_set_aes_cm_256_hmac_sha1_80(crypto_policy_t *p) {
+
+  /*
+   * corresponds to draft-ietf-avt-big-aes-03.txt
+   */
+
+  p->cipher_type     = AES_ICM;           
+  p->cipher_key_len  = 46;
+  p->auth_type       = HMAC_SHA1;             
+  p->auth_key_len    = 20;                /* default 160 bits per RFC 3711 */
+  p->auth_tag_len    = 10;                /* default 80 bits per RFC 3711 */
+  p->sec_serv        = sec_serv_conf_and_auth;
+}
+
+
+void
+crypto_policy_set_aes_cm_256_hmac_sha1_32(crypto_policy_t *p) {
+
+  /*
+   * corresponds to draft-ietf-avt-big-aes-03.txt
+   *
+   * note that this crypto policy is intended for SRTP, but not SRTCP
+   */
+
+  p->cipher_type     = AES_ICM;           
+  p->cipher_key_len  = 46;
+  p->auth_type       = HMAC_SHA1;             
+  p->auth_key_len    = 20;                /* default 160 bits per RFC 3711 */
+  p->auth_tag_len    = 4;                 /* default 80 bits per RFC 3711 */
+  p->sec_serv        = sec_serv_conf_and_auth;
+}
+
+
+/* 
+ * secure rtcp functions
+ */
+
+err_status_t 
+srtp_protect_rtcp(srtp_t ctx, void *rtcp_hdr, int *pkt_octet_len) {
+  srtcp_hdr_t *hdr = (srtcp_hdr_t *)rtcp_hdr;
+  uint32_t *enc_start;      /* pointer to start of encrypted portion  */
+  uint32_t *auth_start;     /* pointer to start of auth. portion      */
+  uint32_t *trailer;        /* pointer to start of trailer            */
+  unsigned enc_octet_len = 0;/* number of octets in encrypted portion */
+  uint8_t *auth_tag = NULL; /* location of auth_tag within packet     */
+  err_status_t status;   
+  int tag_len;
+  srtp_stream_ctx_t *stream;
+  int prefix_len;
+  uint32_t seq_num;
+
+  /* we assume the hdr is 32-bit aligned to start */
+  /*
+   * look up ssrc in srtp_stream list, and process the packet with 
+   * the appropriate stream.  if we haven't seen this stream before,
+   * there's only one key for this srtp_session, and the cipher
+   * supports key-sharing, then we assume that a new stream using
+   * that key has just started up
+   */
+  stream = srtp_get_stream(ctx, hdr->ssrc);
+  if (stream == NULL) {
+    if (ctx->stream_template != NULL) {
+      srtp_stream_ctx_t *new_stream;
+      
+      /* allocate and initialize a new stream */
+      status = srtp_stream_clone(ctx->stream_template,
+				 hdr->ssrc, &new_stream); 
+      if (status)
+	return status;
+      
+      /* add new stream to the head of the stream_list */
+      new_stream->next = ctx->stream_list;
+      ctx->stream_list = new_stream;
+      
+      /* set stream (the pointer used in this function) */
+      stream = new_stream;
+    } else {
+      /* no template stream, so we return an error */
+      return err_status_no_ctx;
+    } 
+  }
+  
+  /* 
+   * verify that stream is for sending traffic - this check will
+   * detect SSRC collisions, since a stream that appears in both
+   * srtp_protect() and srtp_unprotect() will fail this test in one of
+   * those functions.
+   */
+  if (stream->direction != dir_srtp_sender) {
+    if (stream->direction == dir_unknown) {
+      stream->direction = dir_srtp_sender;
+    } else {
+      srtp_handle_event(ctx, stream, event_ssrc_collision);
+    }
+  }  
+
+  /* get tag length from stream context */
+  tag_len = auth_get_tag_length(stream->rtcp_auth); 
+
+  /*
+   * set encryption start and encryption length - if we're not
+   * providing confidentiality, set enc_start to NULL
+   */
+  enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header;  
+  enc_octet_len = *pkt_octet_len - octets_in_rtcp_header;
+
+  /* all of the packet, except the header, gets encrypted */
+  /* NOTE: hdr->length is not usable - it refers to only the first
+	 RTCP report in the compound packet! */
+  /* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
+	 multiples of 32-bits (RFC 3550 6.1) */
+  trailer = (uint32_t *) ((char *)enc_start + enc_octet_len);
+
+  if (stream->rtcp_services & sec_serv_conf) {
+    *trailer = htonl(SRTCP_E_BIT);     /* set encrypt bit */    
+  } else {
+    enc_start = NULL;
+    enc_octet_len = 0;
+	/* 0 is network-order independant */
+    *trailer = 0x00000000;     /* set encrypt bit */    
+  }
+
+  /* 
+   * set the auth_start and auth_tag pointers to the proper locations
+   * (note that srtpc *always* provides authentication, unlike srtp)
+   */
+  /* Note: This would need to change for optional mikey data */
+  auth_start = (uint32_t *)hdr;
+  auth_tag = (uint8_t *)hdr + *pkt_octet_len + sizeof(srtcp_trailer_t); 
+
+  /* perform EKT processing if needed */
+  ekt_write_data(stream->ekt, auth_tag, tag_len, pkt_octet_len, 
+		 rdbx_get_packet_index(&stream->rtp_rdbx));
+
+  /* 
+   * check sequence number for overruns, and copy it into the packet
+   * if its value isn't too big
+   */
+  status = rdb_increment(&stream->rtcp_rdb);
+  if (status)
+    return status;
+  seq_num = rdb_get_value(&stream->rtcp_rdb);
+  *trailer |= htonl(seq_num);
+  debug_print(mod_srtp, "srtcp index: %x", seq_num);
+
+  /* 
+   * if we're using rindael counter mode, set nonce and seq 
+   */
+  if (stream->rtcp_cipher->type->id == AES_ICM) {
+    v128_t iv;
+    
+    iv.v32[0] = 0;
+    iv.v32[1] = hdr->ssrc;  /* still in network order! */
+    iv.v32[2] = htonl(seq_num >> 16);
+    iv.v32[3] = htonl(seq_num << 16);
+    status = cipher_set_iv(stream->rtcp_cipher, &iv);
+
+  } else {  
+    v128_t iv;
+    
+    /* otherwise, just set the index to seq_num */  
+    iv.v32[0] = 0;
+    iv.v32[1] = 0;
+    iv.v32[2] = 0;
+    iv.v32[3] = htonl(seq_num);
+    status = cipher_set_iv(stream->rtcp_cipher, &iv);
+  }
+  if (status)
+    return err_status_cipher_fail;
+
+  /* 
+   * if we're authenticating using a universal hash, put the keystream
+   * prefix into the authentication tag
+   */
+  
+  /* if auth_start is non-null, then put keystream into tag  */
+  if (auth_start) {
+
+    /* put keystream prefix into auth_tag */
+    prefix_len = auth_get_prefix_length(stream->rtcp_auth);    
+    status = cipher_output(stream->rtcp_cipher, auth_tag, prefix_len);
+
+    debug_print(mod_srtp, "keystream prefix: %s", 
+		octet_string_hex_string(auth_tag, prefix_len));
+
+    if (status)
+      return err_status_cipher_fail;
+  }
+
+  /* if we're encrypting, exor keystream into the message */
+  if (enc_start) {
+    status = cipher_encrypt(stream->rtcp_cipher, 
+			    (uint8_t *)enc_start, &enc_octet_len);
+    if (status)
+      return err_status_cipher_fail;
+  }
+
+  /* initialize auth func context */
+  auth_start(stream->rtcp_auth);
+
+  /* 
+   * run auth func over packet (including trailer), and write the
+   * result at auth_tag 
+   */
+  status = auth_compute(stream->rtcp_auth, 
+			(uint8_t *)auth_start, 
+			(*pkt_octet_len) + sizeof(srtcp_trailer_t), 
+			auth_tag);
+  debug_print(mod_srtp, "srtcp auth tag:    %s", 
+	      octet_string_hex_string(auth_tag, tag_len));
+  if (status)
+    return err_status_auth_fail;   
+    
+  /* increase the packet length by the length of the auth tag and seq_num*/
+  *pkt_octet_len += (tag_len + sizeof(srtcp_trailer_t));
+    
+  return err_status_ok;  
+}
+
+
+err_status_t 
+srtp_unprotect_rtcp(srtp_t ctx, void *srtcp_hdr, int *pkt_octet_len) {
+  srtcp_hdr_t *hdr = (srtcp_hdr_t *)srtcp_hdr;
+  uint32_t *enc_start;      /* pointer to start of encrypted portion  */
+  uint32_t *auth_start;     /* pointer to start of auth. portion      */
+  uint32_t *trailer;        /* pointer to start of trailer            */
+  unsigned enc_octet_len = 0;/* number of octets in encrypted portion */
+  uint8_t *auth_tag = NULL; /* location of auth_tag within packet     */
+  uint8_t tmp_tag[SRTP_MAX_TAG_LEN];
+  uint8_t tag_copy[SRTP_MAX_TAG_LEN];
+  err_status_t status;   
+  unsigned auth_len;
+  int tag_len;
+  srtp_stream_ctx_t *stream;
+  int prefix_len;
+  uint32_t seq_num;
+
+  /* we assume the hdr is 32-bit aligned to start */
+  /*
+   * look up ssrc in srtp_stream list, and process the packet with 
+   * the appropriate stream.  if we haven't seen this stream before,
+   * there's only one key for this srtp_session, and the cipher
+   * supports key-sharing, then we assume that a new stream using
+   * that key has just started up
+   */
+  stream = srtp_get_stream(ctx, hdr->ssrc);
+  if (stream == NULL) {
+    if (ctx->stream_template != NULL) {
+      stream = ctx->stream_template;
+
+      /* 
+       * check to see if stream_template has an EKT data structure, in
+       * which case we initialize the template using the EKT policy
+       * referenced by that data (which consists of decrypting the
+       * master key from the EKT field)
+       *
+       * this function initializes a *provisional* stream, and this
+       * stream should not be accepted until and unless the packet
+       * passes its authentication check
+       */ 
+      if (stream->ekt != NULL) {
+	status = srtp_stream_init_from_ekt(stream, srtcp_hdr, *pkt_octet_len);
+	if (status)
+	  return status;
+      }
+
+      debug_print(mod_srtp, "srtcp using provisional stream (SSRC: 0x%08x)", 
+		  hdr->ssrc);
+    } else {
+      /* no template stream, so we return an error */
+      return err_status_no_ctx;
+    } 
+  }
+  
+  /* get tag length from stream context */
+  tag_len = auth_get_tag_length(stream->rtcp_auth); 
+
+  /*
+   * set encryption start, encryption length, and trailer
+   */
+  enc_octet_len = *pkt_octet_len - 
+                  (octets_in_rtcp_header + tag_len + sizeof(srtcp_trailer_t));
+  /* index & E (encryption) bit follow normal data.  hdr->len
+	 is the number of words (32-bit) in the normal packet minus 1 */
+  /* This should point trailer to the word past the end of the
+	 normal data. */
+  /* This would need to be modified for optional mikey data */
+  /*
+   * NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
+   *	 multiples of 32-bits (RFC 3550 6.1)
+   */
+  trailer = (uint32_t *) ((char *) hdr +
+		     *pkt_octet_len -(tag_len + sizeof(srtcp_trailer_t)));
+  if (*((unsigned char *) trailer) & SRTCP_E_BYTE_BIT) {
+    enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header;  
+  } else {
+    enc_octet_len = 0;
+    enc_start = NULL; /* this indicates that there's no encryption */
+  }
+
+  /* 
+   * set the auth_start and auth_tag pointers to the proper locations
+   * (note that srtcp *always* uses authentication, unlike srtp)
+   */
+  auth_start = (uint32_t *)hdr;
+  auth_len = *pkt_octet_len - tag_len;
+  auth_tag = (uint8_t *)hdr + auth_len;
+
+  /* 
+   * if EKT is in use, then we make a copy of the tag from the packet,
+   * and then zeroize the location of the base tag
+   *
+   * we first re-position the auth_tag pointer so that it points to
+   * the base tag
+   */
+  if (stream->ekt) {
+    auth_tag -= ekt_octets_after_base_tag(stream->ekt);
+    memcpy(tag_copy, auth_tag, tag_len);
+    octet_string_set_to_zero(auth_tag, tag_len);
+    auth_tag = tag_copy;
+    auth_len += tag_len;
+  }
+
+  /* 
+   * check the sequence number for replays
+   */
+  /* this is easier than dealing with bitfield access */
+  seq_num = ntohl(*trailer) & SRTCP_INDEX_MASK;
+  debug_print(mod_srtp, "srtcp index: %x", seq_num);
+  status = rdb_check(&stream->rtcp_rdb, seq_num);
+  if (status)
+    return status;
+
+  /* 
+   * if we're using aes counter mode, set nonce and seq 
+   */
+  if (stream->rtcp_cipher->type->id == AES_ICM) {
+    v128_t iv;
+
+    iv.v32[0] = 0;
+    iv.v32[1] = hdr->ssrc; /* still in network order! */
+    iv.v32[2] = htonl(seq_num >> 16);
+    iv.v32[3] = htonl(seq_num << 16);
+    status = cipher_set_iv(stream->rtcp_cipher, &iv);
+
+  } else {  
+    v128_t iv;
+    
+    /* otherwise, just set the index to seq_num */  
+    iv.v32[0] = 0;
+    iv.v32[1] = 0;
+    iv.v32[2] = 0;
+    iv.v32[3] = htonl(seq_num);
+    status = cipher_set_iv(stream->rtcp_cipher, &iv);
+
+  }
+  if (status)
+    return err_status_cipher_fail;
+
+  /* initialize auth func context */
+  auth_start(stream->rtcp_auth);
+
+  /* run auth func over packet, put result into tmp_tag */
+  status = auth_compute(stream->rtcp_auth, (uint8_t *)auth_start,  
+			auth_len, tmp_tag);
+  debug_print(mod_srtp, "srtcp computed tag:       %s", 
+	      octet_string_hex_string(tmp_tag, tag_len));
+  if (status)
+    return err_status_auth_fail;   
+  
+  /* compare the tag just computed with the one in the packet */
+  debug_print(mod_srtp, "srtcp tag from packet:    %s", 
+	      octet_string_hex_string(auth_tag, tag_len));  
+  if (octet_string_is_eq(tmp_tag, auth_tag, tag_len))
+    return err_status_auth_fail;
+
+  /* 
+   * if we're authenticating using a universal hash, put the keystream
+   * prefix into the authentication tag
+   */
+  prefix_len = auth_get_prefix_length(stream->rtcp_auth);    
+  if (prefix_len) {
+    status = cipher_output(stream->rtcp_cipher, auth_tag, prefix_len);
+    debug_print(mod_srtp, "keystream prefix: %s", 
+		octet_string_hex_string(auth_tag, prefix_len));
+    if (status)
+      return err_status_cipher_fail;
+  }
+
+  /* if we're decrypting, exor keystream into the message */
+  if (enc_start) {
+    status = cipher_decrypt(stream->rtcp_cipher, 
+			    (uint8_t *)enc_start, &enc_octet_len);
+    if (status)
+      return err_status_cipher_fail;
+  }
+
+  /* decrease the packet length by the length of the auth tag and seq_num */
+  *pkt_octet_len -= (tag_len + sizeof(srtcp_trailer_t));
+
+  /*
+   * if EKT is in effect, subtract the EKT data out of the packet
+   * length
+   */
+  *pkt_octet_len -= ekt_octets_after_base_tag(stream->ekt);
+
+  /* 
+   * verify that stream is for received traffic - this check will
+   * detect SSRC collisions, since a stream that appears in both
+   * srtp_protect() and srtp_unprotect() will fail this test in one of
+   * those functions.
+   *
+   * we do this check *after* the authentication check, so that the
+   * latter check will catch any attempts to fool us into thinking
+   * that we've got a collision
+   */
+  if (stream->direction != dir_srtp_receiver) {
+    if (stream->direction == dir_unknown) {
+      stream->direction = dir_srtp_receiver;
+    } else {
+      srtp_handle_event(ctx, stream, event_ssrc_collision);
+    }
+  }
+
+  /* 
+   * if the stream is a 'provisional' one, in which the template context
+   * is used, then we need to allocate a new stream at this point, since
+   * the authentication passed
+   */
+  if (stream == ctx->stream_template) {  
+    srtp_stream_ctx_t *new_stream;
+
+    /* 
+     * allocate and initialize a new stream 
+     * 
+     * note that we indicate failure if we can't allocate the new
+     * stream, and some implementations will want to not return
+     * failure here
+     */
+    status = srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream); 
+    if (status)
+      return status;
+    
+    /* add new stream to the head of the stream_list */
+    new_stream->next = ctx->stream_list;
+    ctx->stream_list = new_stream;
+    
+    /* set stream (the pointer used in this function) */
+    stream = new_stream;
+  }
+
+  /* we've passed the authentication check, so add seq_num to the rdb */
+  rdb_add_index(&stream->rtcp_rdb, seq_num);
+    
+    
+  return err_status_ok;  
+}
+
+
+
+/*
+ * dtls keying for srtp 
+ */
+
+err_status_t
+crypto_policy_set_from_profile_for_rtp(crypto_policy_t *policy, 
+				       srtp_profile_t profile) {
+
+  /* set SRTP policy from the SRTP profile in the key set */
+  switch(profile) {
+  case srtp_profile_aes128_cm_sha1_80:
+    crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
+    crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_aes128_cm_sha1_32:
+    crypto_policy_set_aes_cm_128_hmac_sha1_32(policy);
+    crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_null_sha1_80:
+    crypto_policy_set_null_cipher_hmac_sha1_80(policy);
+    crypto_policy_set_null_cipher_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_aes256_cm_sha1_80:
+    crypto_policy_set_aes_cm_256_hmac_sha1_80(policy);
+    crypto_policy_set_aes_cm_256_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_aes256_cm_sha1_32:
+    crypto_policy_set_aes_cm_256_hmac_sha1_32(policy);
+    crypto_policy_set_aes_cm_256_hmac_sha1_80(policy);
+    break;
+    /* the following profiles are not (yet) supported */
+  case srtp_profile_null_sha1_32:
+  default:
+    return err_status_bad_param;
+  }
+
+  return err_status_ok;
+}
+
+err_status_t
+crypto_policy_set_from_profile_for_rtcp(crypto_policy_t *policy, 
+					srtp_profile_t profile) {
+
+  /* set SRTP policy from the SRTP profile in the key set */
+  switch(profile) {
+  case srtp_profile_aes128_cm_sha1_80:
+    crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_aes128_cm_sha1_32:
+    crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_null_sha1_80:
+    crypto_policy_set_null_cipher_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_aes256_cm_sha1_80:
+    crypto_policy_set_aes_cm_256_hmac_sha1_80(policy);
+    break;
+  case srtp_profile_aes256_cm_sha1_32:
+    crypto_policy_set_aes_cm_256_hmac_sha1_80(policy);
+    break;
+    /* the following profiles are not (yet) supported */
+  case srtp_profile_null_sha1_32:
+  default:
+    return err_status_bad_param;
+  }
+
+  return err_status_ok;
+}
+
+void
+append_salt_to_key(uint8_t *key, unsigned int bytes_in_key,
+		   uint8_t *salt, unsigned int bytes_in_salt) {
+
+  memcpy(key + bytes_in_key, salt, bytes_in_salt);
+
+}
+
+unsigned int
+srtp_profile_get_master_key_length(srtp_profile_t profile) {
+
+  switch(profile) {
+  case srtp_profile_aes128_cm_sha1_80:
+    return 16;
+    break;
+  case srtp_profile_aes128_cm_sha1_32:
+    return 16;
+    break;
+  case srtp_profile_null_sha1_80:
+    return 16;
+    break;
+  case srtp_profile_aes256_cm_sha1_80:
+    return 32;
+    break;
+  case srtp_profile_aes256_cm_sha1_32:
+    return 32;
+    break;
+    /* the following profiles are not (yet) supported */
+  case srtp_profile_null_sha1_32:
+  default:
+    return 0;  /* indicate error by returning a zero */
+  }
+}
+
+unsigned int
+srtp_profile_get_master_salt_length(srtp_profile_t profile) {
+
+  switch(profile) {
+  case srtp_profile_aes128_cm_sha1_80:
+    return 14;
+    break;
+  case srtp_profile_aes128_cm_sha1_32:
+    return 14;
+    break;
+  case srtp_profile_null_sha1_80:
+    return 14;
+    break;
+  case srtp_profile_aes256_cm_sha1_80:
+    return 14;
+    break;
+  case srtp_profile_aes256_cm_sha1_32:
+    return 14;
+    break;
+    /* the following profiles are not (yet) supported */
+  case srtp_profile_null_sha1_32:
+  default:
+    return 0;  /* indicate error by returning a zero */
+  }
+}
Property changes on: libsrtp/srtp/srtp.c
___________________________________________________________________
Added: svn:eol-style
   + LF