Ok, so I have calls cutting off after 30 seconds on one type of handset but not another. Looking at ngrep, two things that jump out at me: 1. The working handset is sending ACK responses to FreeSWITCH's SDP messages, but the other one isn't. Presumably this is why FS is issuing BYE and dropping the call after 30 seconds, but there are no settings on the phone to this affect (Siemens Gigaset), and I'm not sure whether it's related to point 2 below: 2. The one which is cutting off is using UDP, and the working one TCP. TBH I can't see why it's using UDP when the SIP profile etc is set to use TCP!? Can anyone give me some pointers / see anything I'm missing? U 2012/07/03 15:40:18.370502 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP (HANDSET_IP):5060;branch=z9hG4bKeb201ee255084c9790fb39182842931b;rport=5060. From: <sip:1002@(FS_PRIVATE_IP)>;tag=2284500536. To: <sip:01234567890@(FS_PRIVATE_IP);user=phone>;tag=graNreK1vcrXg. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 3 INVITE. Contact: <sip:01234567890@(FS_PUBLIC_IP):5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 245. Remote-Party-ID: "Outbound Call" <sip:01234567890@(FS_PRIVATE_IP)>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1341299187 1341299188 IN IP4 (FS_PRIVATE_IP). s=FreeSWITCH. c=IN IP4 (FS_PRIVATE_IP). t=0 0. m=audio 27202 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # T 2012/07/03 15:40:19.841678 (ITSP_IP):5060 -> (FS_PRIVATE_IP):35664 [AP] OPTIONS sip:gw+MyGateway(FS_PUBLIC_IP):36304;tport=tcp;transport=tcp;gw=MyGateway SIP/2.0. Via: SIP/2.0/TCP (ITSP_IP);branch=z9hG4bKaae6e0SZ07p9c. Max-Forwards: 70. From: <sip:mod_sofia@(ITSP_IP):5060>;tag=Fp3vKpcZBN4UF. To: <sip:istp_username@sip.my-itsp.com>. Call-ID: d8f67e63-3fbf-1230-2cb8-001b78596f84. CSeq: 31241096 OPTIONS. Contact: <sip:mod_sofia@(ITSP_IP):5060>. User-Agent: my-itsp.com. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. . # T 2012/07/03 15:40:19.842080 (FS_PRIVATE_IP):35664 -> (ITSP_IP):5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP (ITSP_IP);branch=z9hG4bKaae6e0SZ07p9c;rport=5060. From: <sip:mod_sofia@(ITSP_IP):5060>;tag=Fp3vKpcZBN4UF. To: <sip:istp_username@sip.my-itsp.com>;tag=Q1y1N4BZU22pe. Call-ID: d8f67e63-3fbf-1230-2cb8-001b78596f84. CSeq: 31241096 OPTIONS. Contact: <sip:gw+MyGateway(FS_PUBLIC_IP):5080;tport=tcp;transport=tcp;gw=MyGateway>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. . ## U 2012/07/03 15:40:22.370503 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP (HANDSET_IP):5060;branch=z9hG4bKeb201ee255084c9790fb39182842931b;rport=5060. From: <sip:1002@(FS_PRIVATE_IP)>;tag=2284500536. To: <sip:01234567890@(FS_PRIVATE_IP);user=phone>;tag=graNreK1vcrXg. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 3 INVITE. Contact: <sip:01234567890@(FS_PUBLIC_IP):5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 245. Remote-Party-ID: "Outbound Call" <sip:01234567890@(FS_PRIVATE_IP)>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1341299187 1341299188 IN IP4 (FS_PRIVATE_IP). s=FreeSWITCH. c=IN IP4 (FS_PRIVATE_IP). t=0 0. m=audio 27202 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U 2012/07/03 15:40:22.863571 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 BYE sip:1002@(HANDSET_IP):5060 SIP/2.0. Via: SIP/2.0/UDP (FS_PUBLIC_IP);rport;branch=z9hG4bKp90ccpcU3jUaN. Max-Forwards: 70. From: <sip:01234567890@(FS_PRIVATE_IP);user=phone>;tag=graNreK1vcrXg. To: <sip:1002@(FS_PRIVATE_IP)>;tag=2284500536. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 30327147 BYE. Contact: <sip:01234567890@(FS_PUBLIC_IP):5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. . (FS_PRIVATE_IP)