Changes between Version 16 and Version 17 of SRTP
- Timestamp:
- Aug 26, 2010 3:12:05 PM (14 years ago)
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SRTP
v16 v17 140 140 141 141 To use SRTP transport directly: 142 - Call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#gf4a10278c4586f5239ee7a698aa4a85d pjmedia_transport_srtp_create()] to create the SRTP adapter, giving it the actual media transport instance (such as UDP transport).142 - Call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#gf4a10278c4586f5239ee7a698aa4a85d pjmedia_transport_srtp_create()] to create the SRTP adapter, giving it the actual media transport instance (such as UDP transport). 143 143 - Call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT__SRTP.htm#gb603d4f070665cb1a8038c021a6019e6 pjmedia_transport_srtp_start()] to active SRTP session, giving it both local and remote crypto settings and keys. 144 - Call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#g13bbd87a2d4868229c95711065905cfa pjmedia_transport_attach()] to configure the remote RTP/RTCP addresses and attach your RTP and RTCP callbacks.145 - Call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#g79db5c82268501ec3bf7c1897c0b3626 pjmedia_transport_send_rtp()] and [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT__H.htm#g5fe97ac16287563420950a8b87d247b4 pjmedia_transport_send_rtcp()] to send RTP/RTCP packets.146 - Once you done with your session, call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#g0114fa7e20cb17c645701b2dbda96452 pjmedia_transport_close()] to destroy the SRTP adapter (and optionally the actual transport which is attached to the SRTP adapter, depending on whether ''close_member_tp'' flag is set in the [http://www.pjsip.org/pjmedia/docs/html/structpjmedia__srtp__setting.htm pjmedia_srtp_setting] when creating the SRTP adapter).144 - Call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#g13bbd87a2d4868229c95711065905cfa pjmedia_transport_attach()] to configure the remote RTP/RTCP addresses and attach your RTP and RTCP callbacks. 145 - Call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#g79db5c82268501ec3bf7c1897c0b3626 pjmedia_transport_send_rtp()] and [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#g5fe97ac16287563420950a8b87d247b4 pjmedia_transport_send_rtcp()] to send RTP/RTCP packets. 146 - Once you done with your session, call [http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#g0114fa7e20cb17c645701b2dbda96452 pjmedia_transport_close()] to destroy the SRTP adapter (and optionally the actual transport which is attached to the SRTP adapter, depending on whether ''close_member_tp'' flag is set in the [http://www.pjsip.org/pjmedia/docs/html/structpjmedia__srtp__setting.htm pjmedia_srtp_setting] when creating the SRTP adapter). 147 147 148 148 ---- … … 156 156 New interfaces in media transport are as follows (please consult the PJMEDIA transport documentation for more info): 157 157 158 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#gf4a10278c4586f5239ee7a698aa4a85d media_create()]'''::158 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#gf4a10278c4586f5239ee7a698aa4a85d media_create()]''':: 159 159 This callback is called by application (or PJSUA-LIB) to allow the media transport to add more information in the SDP offer, before the offer is sent to remote. Additionally, for answerer side, this callback allows the media transport to reject the offer before this offer is processed by the SDP negotiator. 160 160 161 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#g8031382904162781e4b58fcee0a62dcd media_start()]'''::161 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#g8031382904162781e4b58fcee0a62dcd media_start()]''':: 162 162 This callback is called after offer and answer are negotiated, and both SDPs are available, and before the media is started. For answerer side, this callback will be called before the answer is sent to remote, to allow media transport to put additional info in the SDP. For offerer side, this callback will be called after SDP answer is received. In this callback, the media transport has the final chance to negotiate/validate the offer and answer before media is really started (and answer is sent, for answerer side). 163 163 164 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#ged068de25cc5cbda27c1b7058597d3b5 media_stop()]'''::164 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#ged068de25cc5cbda27c1b7058597d3b5 media_stop()]''':: 165 165 This callback is called when the media is stopped, to allow the media transport to release its resources. 166 166 167 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT __H.htm#g30e410ca02a4e815346b5e5fa505e7b5 simulate_lost()]'''::167 '''[http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm#g30e410ca02a4e815346b5e5fa505e7b5 simulate_lost()]''':: 168 168 This has nothing to do with SRTP, but since all media transports support this feature (packet loss simulation), we added this as a new interface. 169 169