Merged revisions 336936 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/10

........
  r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
  
  
  Allow Setting Auth Tag Bit length Based on invite or config option
  
  Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
  Curently only 80 bit is supported.
  
  The outgoing invite will use the taglen of the incoming invite preventing
  one-way audio.
  
  (Closes issue ASTERISK-17895)
  
  Review: https://reviewboard.asterisk.org/r/1173/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Gregory Nietsky
2011-09-20 16:56:11 +00:00
parent 14d3f891e0
commit 8493c46308
7 changed files with 36 additions and 13 deletions

View File

@@ -31,6 +31,7 @@
#include <asterisk/rtp_engine.h>
struct sdp_crypto;
struct sip_srtp;
/*! \brief Initialize an return an sdp_crypto struct
*
@@ -51,11 +52,12 @@ void sdp_crypto_destroy(struct sdp_crypto *crypto);
* \param p A valid sdp_crypto struct
* \param attr the a:crypto line from SDP
* \param rtp The rtp instance associated with the SDP being parsed
* \param srtp SRTP structure
*
* \retval 0 success
* \retval nonzero failure
*/
int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp);
int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp);
/*! \brief Generate an SRTP a=crypto offer
@@ -68,7 +70,7 @@ int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_in
* \retval 0 success
* \retval nonzero failure
*/
int sdp_crypto_offer(struct sdp_crypto *p);
int sdp_crypto_offer(struct sdp_crypto *p, int taglen);
/*! \brief Return the a_crypto value of the sdp_crypto struct