res_rtp_asterisk / res_pjsip: Add support for BUNDLE.
BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11
Showing
- channels/chan_pjsip.c 0 additions, 2 deletionschannels/chan_pjsip.c
- include/asterisk/res_pjsip.h 2 additions, 0 deletionsinclude/asterisk/res_pjsip.h
- include/asterisk/res_pjsip_session.h 19 additions, 0 deletionsinclude/asterisk/res_pjsip_session.h
- include/asterisk/rtp_engine.h 52 additions, 0 deletionsinclude/asterisk/rtp_engine.h
- main/rtp_engine.c 58 additions, 10 deletionsmain/rtp_engine.c
- res/res_pjsip.c 8 additions, 0 deletionsres/res_pjsip.c
- res/res_pjsip/pjsip_configuration.c 5 additions, 0 deletionsres/res_pjsip/pjsip_configuration.c
- res/res_pjsip_sdp_rtp.c 294 additions, 117 deletionsres/res_pjsip_sdp_rtp.c
- res/res_pjsip_session.c 211 additions, 2 deletionsres/res_pjsip_session.c
- res/res_pjsip_t38.c 10 additions, 1 deletionres/res_pjsip_t38.c
- res/res_rtp_asterisk.c 495 additions, 245 deletionsres/res_rtp_asterisk.c
Loading
Please register or sign in to comment