Re: [asterisk-users] SIP OPTIONS storm?
- Original Message - On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote: I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Just because Box B was receiving 65MBps doesn’t mean box A was sending them. I suspect it’s probably the same one repeated, due to some kind of network problem. Do you have a pcap so you can look for the ID in the packets to see if they are the same? Would be good if you can prove A sent them too (traffic stats from SNMP monitoring or something). Right, but a packet capture shows the source to be box A, and the destination to be box B. NMS reports from the same time period confirm the traffic flows. I'm not guessing or stabbing in the dark, I did my homework before posting. :) Checking the IDs across ~25 packets, all have different SIP IDs. Any thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
Attach the packet capture to your Jira bug report or post it online somewhere. Hopefully someone will look at it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Tuesday, February 18, 2014 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP OPTIONS storm? - Original Message - On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote: I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Just because Box B was receiving 65MBps doesn’t mean box A was sending them. I suspect it’s probably the same one repeated, due to some kind of network problem. Do you have a pcap so you can look for the ID in the packets to see if they are the same? Would be good if you can prove A sent them too (traffic stats from SNMP monitoring or something). Right, but a packet capture shows the source to be box A, and the destination to be box B. NMS reports from the same time period confirm the traffic flows. I'm not guessing or stabbing in the dark, I did my homework before posting. :) Checking the IDs across ~25 packets, all have different SIP IDs. Any thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
- Original Message - SIP options message is due to check the peer registration is keepalive. As per my understanding it might be because of network flap may be wireshark trace can give you any clue. Regards Correct. I understand the role and function of the OPTIONS requests. The issue is why was Asterisk sending out 65Mbps worth of them to one peer? I did get a capture of the traffic, but nothing appears to explain *why* the traffic was there to begin with. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote: I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Just because Box B was receiving 65MBps doesn’t mean box A was sending them. I suspect it’s probably the same one repeated, due to some kind of network problem. Do you have a pcap so you can look for the ID in the packets to see if they are the same? Would be good if you can prove A sent them too (traffic stats from SNMP monitoring or something). S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
SIP options message is due to check the peer registration is keepalive. As per my understanding it might be because of network flap may be wireshark trace can give you any clue. Regards On 13 Feb 2014 23:41, Tim Nelson tnel...@rockbochs.com wrote: Greetings- I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Of course, logs on Box A were not set to show debug info, so there is no indication of a problem. Logs on Box B show no issues, only at a very specific start time, there are suddenly tons of: [2014-02-13 00:12:50] DEBUG[31516] chan_sip.c: Allocating new SIP dialog for 2a338cf5518531e31190bd4b7826d137@x.y.z.166:5060 - OPTIONS (No RTP) I've done quite a bit of searching, but am not finding anything of consequence. Also, the Asterisk changelogs are not providing anything that would indicate this is known and fixed, at least for the 11.x branch. Thoughts/suggestions? Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
