Re: [SR-Users] rtpengine - SRTP <> RTP missing a=crypto

Hi Everyone, I have Kamailio sitting between MS Teams and Asterisk, and using rtpengine to terminate SRTP on Kamailio so that all my internal traffic is unencrypted. My current config works fine for inbound calls where I initiate the INVITE and Teams responds, but if Teams sends the INVITE I am having an issue where SRTP cannot finish negotiating. Non SRTP calls work fine with RTPEngine as well, so it’s just the RTP to SRTP I am struggling with. According to this I believe I must pass a=crypto in response to the INVITE which also has a=crypto: https://www.dialogic.com/-/media/1f8b54b43087407d9c2b38846c5c2cb5.ashx?h=40… <https://www.dialogic.com/-/media/1f8b54b43087407d9c2b38846c5c2cb5.ashx?h=408&w=622> You can see that in the initial invite from Teams, I get RTP/SAVP with a=crypto, but I do not send one in my OK response after 183 Session In Progress. *As below – I am wondering if it’s because not all audio channels seem to be getting swapped to SAVP?* I’d like to do a generic SRTP <> RTP bridge config (I’ve tried below). However, I am not 100% sure on how to detect when to swap between AVP and SAVP, so I’ve also tried just doing rtpengine_manage() and relying on other code to swap between SAVP or AVP **only** when going to/from Teams to keep it simple. I also tried both with and without “replace-origin replace-session-connection ICE=remove” but I still get the same behaviour in all cases. Any advice appreciated, as this is my first time dealing with SRTP (and rtpengine). Feeling very stuck. Thanks! branch_route[MANAGE_BRANCH] { …         route(NATMANAGE);         route(HANDLE_SRTP); } onreply_route[MANAGE_REPLY] {         xdbg("incoming reply\n");         if(status=~"[12][0-9][0-9]") {                 route(NATMANAGE);         }         route(HANDLE_SRTP); } route[HANDLE_SRTP] {         if (!has_body("application/sdp")) {                 return;         }         rtpengine_manage();         return; # As a test, just do rtpengine_manage() and set SAVP/AVP elsewhere. Same behaviour.         # Handle bridging of RTP and SRTP         # Inbound traffic to SBC should be converted from SRTP to RTP         if (proto==TLS) { rtpengine_manage("RTP/AVP");         # Outbound traffic destined to a TLS destination should be converted from RTP to SRTP         } else if ($ru =~ "transport=tls") { rtpengine_manage("RTP/SAVP");         } } # INVITE from teams rtpengine_manage("replace-origin replace-session-connection ICE=remove  RTP/AVP"); # INVITE to teams rtpengine_manage("replace-origin replace-session-connection ICE=remove  RTP/SAVP"); INVITE sip:+614xxxx@rh.sbc-syd-01.teams.xxxx:5061;user=phone;transport=tls SIP/2.0^M … v=0^M o=- 57931 0 IN IP4 127.0.0.1^M s=session^M c=IN IP4 52.113.76.53^M b=CT:10000000^M t=0 0^M *m=audio 51398 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118^M* c=IN IP4 52.113.76.53^M a=rtcp:51399^M a=ice-ufrag:C8ss^M a=ice-pwd:2bV9D6GcXF5f8m0px/wufQD/^M a=rtcp-mux^M a=candidate:1 1 UDP 2130706431 52.113.76.53 51398 typ srflx raddr 10.0.32.179 rport 51398^M a=candidate:1 2 UDP 2130705918 52.113.76.53 51399 typ srflx raddr 10.0.32.179 rport 51399^M a=candidate:2 1 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=candidate:2 2 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=label:main-audio^M a=mid:1^M *a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:geUHLB1mshmnI5hN83bnO57Hbdm2i7dD14sDAnpA|2^31^M* a=sendrecv^M a=rtpmap:104 SILK/16000^M a=rtpmap:9 G722/8000^M a=rtpmap:103 SILK/8000^M a=rtpmap:111 SIREN/16000^M a=fmtp:111 bitrate=16000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:97 RED/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=rtpmap:13 CN/8000^M a=rtpmap:118 CN/16000^M a=ptime:20^M I correctly convert to/from RTP/AVP and RTP/SAVP for the 183 Session in progress. It is RTP/SAVP before going to Teams: SIP/2.0 183 Session Progress^M … v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M *m=audio 37820 RTP/SAVP 9 8 0 101^M* a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M *m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M* But then when I send the OK after the 183, I am setting RTP/SAVP before sending to MS Teams, but not setting a=crypto: *Also note that I can see there are _/some/_ channels still as RTP/AVP so maybe this is part of the issue.* SIP/2.0 200 OK^M … v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M *m=audio 37820 RTP/SAVP 9 8 0 101^M *a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M *m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M* *Rhys Hanrahan* | Chief Information Officer *e:* rhys(a)nexusone.com.au <mailto:rhys@nexusone.com.au> www.nexusone.com.au <http://www.nexusone.com.au/>signature_1116663581 <http://www.fusiontech.com.au/> * **NEXUS ONE****|** FUSION TECHNOLOGY SOLUTIONS **p:* 1800 NEXUS1 (1800 639 871) or 1800 565 845 *|* *a:* Suite 12.03 Level 12, 227 Elizabeth Street, Sydney NSW 2000 www.nexusone.com.au <http://www.nexusone.com.au/> *|* www.fusiontech.com.au <http://www.fusiontech.com.au/> /The information in this email and any accompanying attachments may contain; a. 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