Cisco invite w/ out SDP. Result is one way audio. Asterisk user can hear remote. Asterisk does not send any SRTP/RTP. <--- SIP read from TLS:(((Cisco_IP))):49416 ---> INVITE sip:3142702001@dsn.mil:5061;maddr=(((Asterisk_IP)));transport=tls;user=phone;trusted;gw;inter-enclave SIP/2.0 From: "Cisco - 7961G";tag=476BE8A0-1108 To: Call-ID: 640599d4d2e8514e1745ef975052df16facd4de54@(((Cisco_IP))) CSeq: 101 INVITE Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba85f-6c09b615-4f13acd2 Contact: User-Agent: Cisco-SIPGateway/IOS-12.x Expires: 180 Max-Forwards: 66 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,timer,resource-priority,replaces,sdp-anat Remote-Party-ID: ;screen=yes;party=calling P-Asserted-Identity: Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,INFO,REGISTER x-nt-corr-id: 140599d4d28f283745ef975db0caaf1df32c334@(((Cisco_IP))) x-nt-location: 2062 Allow-Events: telephone-event Date: Mon, 03 Jan 2011 20:24:29 GMT Resource-Priority: dsn-000000.0 Timestamp: 1294086269 History-Info: ;index=1, P-Called-Party-Id: Session-Expires: 1800 Min-SE: 1800 Cisco-Guid: 0744859520-3525389908-0000000370-0352390505 Content-Length: 0 <-------------> <--- Transmitting (no NAT) to (((Cisco_IP))):5061 ---> SIP/2.0 422 Session Interval Too Small Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba85f-6c09b615-4f13acd2;received=(((Cisco_IP))) From: "Cisco - 7961G";tag=476BE8A0-1108 To: ;tag=as30e74823 Call-ID: 640599d4d2e8514e1745ef975052df16facd4de54@(((Cisco_IP))) CSeq: 101 INVITE Server: VoiceWise Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, UPDATE Supported: replaces, timer, sdp-anat, 100rel Date: Mon, 03 Jan 2011 20:37:19 GMT Min-SE: 180000 Content-Length: 0 <------------> Scheduling destruction of SIP dialog '640599d4d2e8514e1745ef975052df16facd4de54@(((Cisco_IP)))' in 32000 ms (Method: INVITE) <--- SIP read from TLS:(((Cisco_IP))):49416 ---> ACK sip:3142702001@dsn.mil:5061;maddr=(((Asterisk_IP)));transport=tls;user=phone;trusted;gw;inter-enclave SIP/2.0 From: "Cisco - 7961G";tag=476BE8A0-1108 To: ;tag=as30e74823 Call-ID: 640599d4d2e8514e1745ef975052df16facd4de54@(((Cisco_IP))) CSeq: 101 ACK Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba85f-6c09b615-4f13acd2 Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from TLS:(((Cisco_IP))):49416 ---> INVITE sip:3142702001@dsn.mil:5061;maddr=(((Asterisk_IP)));transport=tls;user=phone;trusted;gw;inter-enclave SIP/2.0 From: "Cisco - 7961G";tag=476BE994-12A4 To: Call-ID: 931599d4d20c76b9745ef975ce63ead5c3482876@(((Cisco_IP))) CSeq: 101 INVITE Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba860-6c09b707-14707a9c Contact: User-Agent: Cisco-SIPGateway/IOS-12.x Expires: 180 Max-Forwards: 66 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,timer,resource-priority,replaces,sdp-anat Remote-Party-ID: ;screen=yes;party=calling P-Asserted-Identity: Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,INFO,REGISTER x-nt-corr-id: 431599d4d276a6d7745ef975924c6b18ab6a9943@(((Cisco_IP))) x-nt-location: 2062 Allow-Events: telephone-event Date: Mon, 03 Jan 2011 20:24:29 GMT Resource-Priority: dsn-000000.0 Timestamp: 1294086269 History-Info: ;index=1, P-Called-Party-Id: Session-Expires: 180000 Min-SE: 180000 Cisco-Guid: 0774859520-3525389911-0000000371-0352390505 Content-Length: 0 <-------------> <--- Transmitting (no NAT) to (((Cisco_IP))):5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba860-6c09b707-14707a9c;received=(((Cisco_IP))) From: "Cisco - 7961G";tag=476BE994-12A4 To: Call-ID: 931599d4d20c76b9745ef975ce63ead5c3482876@(((Cisco_IP))) CSeq: 101 INVITE Timestamp: 1294086269 1301883.00 Server: VoiceWise Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, UPDATE Supported: replaces, timer, sdp-anat, 100rel Session-Expires: 180000;refresher=uas Require: timer Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to (((Cisco_IP))):5061 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba860-6c09b707-14707a9c;received=(((Cisco_IP))) From: "Cisco - 7961G";tag=476BE994-12A4 To: ;tag=as23fb60cd Call-ID: 931599d4d20c76b9745ef975ce63ead5c3482876@(((Cisco_IP))) CSeq: 101 INVITE Server: VoiceWise Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, UPDATE Supported: replaces, timer, sdp-anat, 100rel Session-Expires: 180000;refresher=uas Require: timer Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 1034431801 1034431801 IN IP4 (((Asterisk_IP))) s=EdgeAccess t=0 0 m=audio 11498 RTP/AVP 0 8 101 c=IN IP4 (((Asterisk_IP))) a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- Reliably Transmitting (no NAT) to (((Cisco_IP))):5061 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba860-6c09b707-14707a9c;received=(((Cisco_IP))) From: "Cisco - 7961G";tag=476BE994-12A4 To: ;tag=as23fb60cd Call-ID: 931599d4d20c76b9745ef975ce63ead5c3482876@(((Cisco_IP))) CSeq: 101 INVITE Server: VoiceWise Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, UPDATE Supported: replaces, timer, sdp-anat, 100rel Session-Expires: 180000;refresher=uas Require: timer Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 1034431801 1034431802 IN IP4 (((Asterisk_IP))) s=EdgeAccess t=0 0 m=audio 11498 RTP/AVP 0 8 101 c=IN IP4 (((Asterisk_IP))) a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from TLS:(((Cisco_IP))):49416 ---> ACK sip:3142702001@(((Asterisk_IP))):5061;transport=TLS;cca-id=ULSCMFTH10 SIP/2.0 From: "Cisco - 7961G";tag=476BE994-12A4 To: ;tag=as23fb60cd Call-ID: 931599d4d20c76b9745ef975ce63ead5c3482876@(((Cisco_IP))) CSeq: 101 ACK Via: SIP/2.0/TLS (((Cisco_IP))):5061;branch=z9hG4bK-1ba878-6c0a16c4-acafbbe Content-Type: application/sdp Contact: User-Agent: Avaya Aura AS5300 13.0.0.5 Max-Forwards: 67 Allow-Events: telephone-event Date: Mon, 03 Jan 2011 20:24:29 GMT Content-Disposition: session;handling=required Content-Length: 208 v=0 o=CiscoSystemsCCM-SIP 1203 1 IN IP4 (((Cisco_RTP_IP))) s=SIP Call c=IN IP4 (((Cisco_RTP_IP))) t=0 0 m=audio 65196 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 [Jan 3 13:37:44] WARNING[25250]: chan_sip.c:8785 process_sdp: Matched device setup to use SRTP, but request was not! vw9600*CLI>