Andrew Martin
2015-May-08 22:12 UTC
[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
Hello, I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this happening in the asterisk console: http://pastebin.com/7LDwHAJe This problem only happens a fraction of the time, so I have been unable to enable SIP debugging before it happens to get a capture. However, usually the caller will just call back immediately and then the call will work without a problem. It sounds like SIP Timer B is what causes the call to be dropped if an ACK to the INVITE is not received within 32 seconds. How can I determine if this is the case and how can I resolve this "Retransmission timeout" problem? Here is my sip.conf: general] directmedia=no directrtpsetup=no dtmfmode=rfc2833 context=internal allowsubscribe=no qualify=no disallow=all allow=ulaw allow=alaw allow=gsm localnet=10.10.32.0/255.255.248.0 [123] secret=111111 host=dynamic type=friend Thanks! Andrew Martin
Andrew Martin
2015-May-11 17:26 UTC
[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
----- Original Message -----> From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Friday, May 8, 2015 5:12:28 PM > Subject: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Hello, > > I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All > the SIP clients are on a LAN, so no NAT is involved. I have been experiencing > an intermittent problem where a call will be successfully answered, but then > dropped by Asterisk 32 seconds after it is answered (with a "Retransmission > timeout reached on transmission" error). Here is an example of this happening > in the asterisk console: > http://pastebin.com/7LDwHAJe > > This problem only happens a fraction of the time, so I have been unable to > enable SIP debugging before it happens to get a capture. However, usually the > caller will just call back immediately and then the call will work without a > problem. It sounds like SIP Timer B is what causes the call to be dropped if > an > ACK to the INVITE is not received within 32 seconds. How can I determine if > this is the case and how can I resolve this "Retransmission timeout" problem? > > Here is my sip.conf: > general] > directmedia=no > directrtpsetup=no > dtmfmode=rfc2833 > context=internal > allowsubscribe=no > qualify=no > disallow=all > allow=ulaw > allow=alaw > allow=gsm > localnet=10.10.32.0/255.255.248.0 > > > [123] > secret=111111 > host=dynamic > type=friend >By doing a number of test calls today, I have managed to reproduce this while sip debugging was on, so I have that information available now as well: http://pastebin.com/ZJqzdvY3 This was a call from 113 to 146 via a queue. Note that the asterisk server is at 10.10.32.251. I see the following: INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 SIP/2.0 180 Ringing SIP/2.0 180 Ringing SIP/2.0 200 OK ACK sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 SIP/2.0 200 OK ACK sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 This appears to start out with a successful SIP conversation (ending with the first ACK), so it is unclear to me why we have two new sets of INVITEs sent afterwards. Also in case it is relevant, the asterisk server has two NICs set up in a bond with bond-mode 1 (active/backup). Does this additional debug information provide any clues to why this intermittent "retransmission timeout" error is occurring? Thanks, Andrew
Joshua Colp
2015-May-11 17:32 UTC
[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
Andrew Martin wrote:> ----- Original Message -----<snip>> > By doing a number of test calls today, I have managed to reproduce this while > sip debugging was on, so I have that information available now as well: > http://pastebin.com/ZJqzdvY3 > > This was a call from 113 to 146 via a queue. Note that the asterisk server is > at 10.10.32.251. I see the following: > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > SIP/2.0 180 Ringing > SIP/2.0 180 Ringing > SIP/2.0 200 OK > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > SIP/2.0 200 OK > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > > This appears to start out with a successful SIP conversation (ending with the > first ACK), so it is unclear to me why we have two new sets of INVITEs sent > afterwards.Asterisk has sent a re-INVITE to have the media flow directly. The device (seems) to respond with the 200 OK (you can tell based on the CSeq) for the initial INVITE, and not for the re-INVITE. As Asterisk gets no response to its re-INVITE it gives up and terminates the dialog. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
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