Review Board 1.7.16


WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

Review Request #3686 - Created June 28, 2014 and submitted

Joshua Colp
Reviewers
asterisk-dev
Asterisk
This change requires the work for 11 but the merge up is pretty much clean, so this review only includes the PJSIP parts.

SHA-256 support has been added as has two options. One option controls the outgoing transport in SDP offers and the other controls the transport in SDP answers.
Calling.

Diff revision 1

This is not the most recent revision of the diff. The latest diff is revision 3. See what's changed.

1 2 3
1 2 3

  1. /branches/12/include/asterisk/res_pjsip.h: Loading...
  2. /branches/12/include/asterisk/res_pjsip_session.h: Loading...
  3. /branches/12/res/res_pjsip.c: Loading...
  4. /branches/12/res/res_pjsip_sdp_rtp.c: Loading...
  5. /branches/12/res/res_pjsip/pjsip_configuration.c: Loading...
/branches/12/include/asterisk/res_pjsip.h
Revision 417479 New Change
[20] 473 lines
[+20] [+] struct ast_sip_media_rtp_configuration {
474
	unsigned int ice_support;
474
	unsigned int ice_support;
475
	/*! Whether to use the "ptime" attribute received from the endpoint or not */
475
	/*! Whether to use the "ptime" attribute received from the endpoint or not */
476
	unsigned int use_ptime;
476
	unsigned int use_ptime;
477
	/*! Do we use AVPF exclusively for this endpoint? */
477
	/*! Do we use AVPF exclusively for this endpoint? */
478
	unsigned int use_avpf;
478
	unsigned int use_avpf;

    
   
479
	/*! Do we force AVP, AVPF, SAVP, or SAVPF even for DTLS media streams? */

    
   
480
	unsigned int force_avp;

    
   
481
	/*! Do we use the received media transport in our answer SDP */

    
   
482
	unsigned int use_received_transport;
479
	/*! \brief DTLS-SRTP configuration information */
483
	/*! \brief DTLS-SRTP configuration information */
480
	struct ast_rtp_dtls_cfg dtls_cfg;
484
	struct ast_rtp_dtls_cfg dtls_cfg;
481
	/*! Should SRTP use a 32 byte tag instead of an 80 byte tag? */
485
	/*! Should SRTP use a 32 byte tag instead of an 80 byte tag? */
482
	unsigned int srtp_tag_32;
486
	unsigned int srtp_tag_32;
483
	/*! Do we use media encryption? what type? */
487
	/*! Do we use media encryption? what type? */
[+20] [20] 1461 lines
/branches/12/include/asterisk/res_pjsip_session.h
Revision 417479 New Change
 
/branches/12/res/res_pjsip.c
Revision 417479 New Change
 
/branches/12/res/res_pjsip_sdp_rtp.c
Revision 417479 New Change
 
/branches/12/res/res_pjsip/pjsip_configuration.c
Revision 417479 New Change
 
  1. /branches/12/include/asterisk/res_pjsip.h: Loading...
  2. /branches/12/include/asterisk/res_pjsip_session.h: Loading...
  3. /branches/12/res/res_pjsip.c: Loading...
  4. /branches/12/res/res_pjsip_sdp_rtp.c: Loading...
  5. /branches/12/res/res_pjsip/pjsip_configuration.c: Loading...

https://reviewboard.asterisk.org/ runs on a server provided by Digium, Inc. and uses bandwidth donated to the open source Asterisk community by API Digital Communications in Huntsville, AL USA.
Please report problems with this site to asteriskteam@digium.com.