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amartin at xes-inc.com Guest
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Posted: Fri May 08, 2015 5:12 pm Post subject: [asterisk-users] "Retransmission Timeout" results |
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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
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davies147 at gmail.com Guest
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Posted: Wed May 13, 2015 11:39 am Post subject: [asterisk-users] "Retransmission Timeout" results |
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Hi,
In my experience, all Yealink phones work just fine with Asterisk, we have hundreds (perhaps even low-thousands) out there with customers on Asterisk 1.2, 1.6.2, 1.8 and 11.
If you are accurately representing the SIP trace on the phone and the SIP trace on Asterisk, then I would strongly suggest a SIP ALG exists in the network between the two devices and that SIP ALG does not understand SIP properly. The two halves simply do not match, so something must surely be interfering.
In my experience it is often an innocent looking Cisco router. Cisco's SIP implementation is "SIP By Cisco" rather than "RFC compliant SIP". If that is the case Cisco call it a "SIP fixup" and you just need to disable it.
Hope that helps,
Steve
On Wed, 13 May 2015 at 16:59 Andrew Martin <amartin@xes-inc.com (amartin@xes-inc.com)> wrote:
Quote: |
----- Original Message -----
Quote: | From: "Joshua Colp" <jcolp@digium.com (jcolp@digium.com)>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Sent: Wednesday, May 13, 2015 10:50:02 AM
Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
Andrew Martin wrote:
Quote: | Since some packet loss is a possibility, I assume the protocol has
mechanisms
for dealing with it. What should be happening differently in the
communication
when packet loss occurs? Should the phone just be re-sending the OK,
instead of
printing "<0> | ERROR | receive a request with same cseq??" to its log? Or
should
Asterisk be starting with a new cseq on each INVITE retry?
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The 200 OK should be retransmitted until an ACK is received. It honestly
looks like the phone can't talk to Asterisk and it's just generally
screwing up signaling.
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Thanks for the clarification and help debugging this problem. I will work
with the phone vendor to see if they can resolve this from their end. If you
have any other ideas about how to disable re-INVITEs on the asterisk side,
beyond what I have done already, please let me know.
Thanks,
Andrew
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amartin at xes-inc.com Guest
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Posted: Wed May 13, 2015 1:20 pm Post subject: [asterisk-users] "Retransmission Timeout" results |
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----- Original Message -----
Quote: | From: "Steve Davies" <davies147@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Sent: Wednesday, May 13, 2015 11:39:29 AM
Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
Hi,
In my experience, all Yealink phones work just fine with Asterisk, we have
hundreds (perhaps even low-thousands) out there with customers on Asterisk
1.2, 1.6.2, 1.8 and 11.
If you are accurately representing the SIP trace on the phone and the SIP
trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
network between the two devices and that SIP ALG does not understand SIP
properly. The two halves simply do not match, so something must surely be
interfering.
In my experience it is often an innocent looking Cisco router. Cisco's SIP
implementation is "SIP By Cisco" rather than "RFC compliant SIP". If that is
the case Cisco call it a "SIP fixup" and you just need to disable it.
Hope that helps,
Steve
| Steve,
That is an interesting point - the server and the phone are both connected to
Netgear switches where I have enabled their "Auto-VoIP" feature, which remarks
packets based on protocol (SIP, SCCP, etc) for better QoS:
http://kb.netgear.com/app/answers/detail/a_id/21758
I wonder if this remarking process is modifying another part of the packet too?
Both devices are on the same subnet, so although these switches do route
traffic as well, that shouldn't be coming into play here.
Andrew
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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