Sporadic outbound call drops and dynamic IP address updates
Hi, Running freeswitch 1.2.12, built on Debian 7.0. My ext-rtp-ip and ext-sip-ip are set to the freeswitch stun server. The DSL modem has a dynamic IP address, and I'm not currently using a dynamic dns updater (switching to a static IP in a few weeks). I've been having a sporadic issue where outbound calls only last for 25 seconds or so, and are then dropped. The Nexvortex tech said that it was a problem with my FS pbx--that it was ignoring the Contact header and continuing to send to the proxy (66.23.129.253 in this example). I initially thought it might be some problem with the 1.3 version of freeswitch I was using, so I rebuilt using the latest stable (described above). The problem seemed to go away...for a day or so. It was back again today, and no changes were made in the interim. While I was stumbling through trying to figure out what was wrong, magically the outbound calls started behaving again. Again, I didn't touch the configuration...I was just making calls and trying to make sense of the debug info I was seeing (via wireshark and sip trace). I am fairly new to debugging SIP at this level. Below are some excerpts from the sip trace...this was a *successful* outbound call (no drop after 25 seconds). My question is: Is there any merit to what the tech said? I see in this sequence that NV sets the Contact header, but FS continues to send to the 66.23 address, though I also see in the trace that the Contact address is contained in the FS response. I also notice NV seems to be using Sonus, though I didn't see anything on the "RTP Issues" page which seemed relevant. Does anyone have any thoughts about why sometimes everything works ok and other times it doesn't? I have a second FS box (identical platform) registered with a different provider...zero problems. A second, but less pressing question: does FS automatically handle changes to a dynamic external IP address? As I'm using stun, it seems like it should, but the modem received a new address midday a few days ago, and I could no longer receive calls. A "sofia status" revealed that the external profile still showed the old external IP (and this was several hours later). I just restarted FS and everything was fine, but was wondering if I have some config parameter wrong somewhere that it didn't update on its own. Thanks for any help... Tim I fuzzed my phone #s, IPs, and some Proxy Auth info, but everything else should be as logged. recv 874 bytes from udp/[66.23.129.253]:5060 at 02:57:16.514012: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.2:5080;rport=5080;branch=z9hG4bKF9DB405pNct2S Record-Route: <sip:66.23.129.254:5060;lr>,<sip:15136852816@66.23.129.253:5060;nat=y es;ftag=r011rpa0Syg0S;lr=on> From: "Kitchen Angels" <sip:15054717780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400454 INVITE Contact: <sip:72.15.219.140;did=27d.e851d0a3> Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 235 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 17065 29458 IN IP4 208.79.54.210 s=SIP Media Capabilities c=IN IP4 208.79.54.212 t=0 0 m=audio 18928 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 recv 1034 bytes from udp/[66.23.129.253]:5060 at 02:57:21.804773: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5080;rport=5080;branch=z9hG4bKF9DB405pNct2S Record-Route: <sip:66.23.129.254:5060;lr>,<sip:15136852816@66.23.129.253:5060;nat=y es;ftag=r011rpa0Syg0S;lr=on> From: "Bob" <sip:15136857780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400454 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay , multipart/mixed Contact: <sip:72.15.219.140;did=27d.e851d0a3> Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Require: timer Supported: timer Session-Expires: 1800;refresher=uac Content-Length: 235 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 17065 29458 IN IP4 208.79.54.210 s=SIP Media Capabilities c=IN IP4 208.79.54.212 t=0 0 m=audio 18928 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 send 774 bytes to udp/[66.23.129.253]:5060 at 02:57:21.811091: ------------------------------------------------------------------------ ACK sip:72.15.219.140;did=27d.e851d0a3 SIP/2.0 Via: SIP/2.0/UDP 175.29.62.55:5080;rport;branch=z9hG4bKgj735UptjNgNN Route: <sip:15136852816@66.23.129.253:5060;nat=yes;ftag=r011rpa0Syg0S;lr=on> Route: <sip:66.23.129.254:5060;lr> Max-Forwards: 70 From: "Bob" <sip:15136857780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400454 ACK Contact: <sip:gw+nexvortex-main@175.29.62.55:5080;transport=udp;gw=nexvortex-main> Proxy-Authorization: Digest username="dummy", realm="nexvortex.com", nonce="U jkXvFI5FpDt+8Hzk/viti", algorithm=MD5, uri="sip:15136852816@px3.nexvortex.co m", response="20200d7ccb42e873a3f4e16c8f9d9de9" Content-Length: 0 send 1033 bytes to udp/[66.23.129.253]:5060 at 02:58:13.951291: ------------------------------------------------------------------------ BYE sip:72.15.219.140;did=27d.e851d0a3 SIP/2.0 Via: SIP/2.0/UDP 175.29.62.55:5080;rport;branch=z9hG4bKHU0v7p7XFy67g Route: <sip:15136852816@66.23.129.253:5060;nat=yes;ftag=r011rpa0Syg0S;lr=on> Route: <sip:66.23.129.254:5060;lr> Max-Forwards: 70 From: "Bob" <sip:15136857780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400455 BYE Contact: <sip:gw+nexvortex-main@175.29.62.55:5080;transport=udp;gw=nexvortex-main> User-Agent: FreeSWITCH-mod_sofia/1.2.12+git~20130826T225725Z~4990a541bf Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, N OTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="dummy", realm="nexvortex.com", nonce="U jkXvFI5FpDt+8Hzk/viti", algorithm=MD5, uri="sip:72.15.219.140;did=27d.e851d0 a3", response="3561a42f5bfa281fad3f39077534732d" Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 -- Tim Bock Synacktics, LLC www.synacktics.com
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Tim Bock