What is the problem with SIP retransmits?
-----------------------------------------
Sometimes you get messages in the console like these:
- "retrans_pkt: Hanging up call XX77yy - no reply to our critical
packet."
- "retrans_pkt: Cancelling retransmit of OPTIONs"
The SIP protocol is based on requests and replies. Both sides send
requests and wait for replies. Some of these requests are important.
In a TCP/IP network many things can happen with IP packets. Firewalls,
NAT devices, Session Border Controllers and SIP Proxys are in the
signalling path and they will affect the call.
And
What can I do?
--------------
Turn on SIP debug, try to understand the signalling that happens
and see if you're missing the reply to the INVITE or if the
ACK gets lost. When you know what happens, you've taken the
first step to track down the problem. See the list above and
investigate your network.
For NAT and Firewall problems, there are many documents
to help you. Start with reading sip.conf.sample that is
part of your Asterisk distribution.
The SIP signalling standard, including retransmissions
and timers for these, is well documented in the IETF
RFC 3261.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Todd Reese
Sent: Friday, September 19, 2008 3:54 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Dropping Phone Calls
Hi All,
I'm currently having trouble with dropped phone calls. The following error
message is always in the log. This is a Grandstream GXP-2000 Firmware
1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been
occurring on other versions also.
[Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum
retries exceeded on transmission 8acaea6dc4c6e9b5 at 10.11.17.23 for seqno
50706 (Critical Response) -- See doc/sip-retransmit.txt.
[Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1980 retrans_pkt: Hanging up
call 8acaea6dc4c6e9b5 at 10.11.17.23 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Any Ideas?
Regards,
Todd Reese
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<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>From the
“doc/sip-retransmit.txt”<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>What is the problem with SIP
retransmits?<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>-----------------------------------------<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sometimes you get messages in the console like
these:<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>- "retrans_pkt: Hanging up call XX77yy
- no reply to our
critical packet."<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>- "retrans_pkt: Cancelling retransmit of
OPTIONs"<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The SIP protocol is based on requests and replies. Both
sides
send<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>requests and wait for replies. Some of these requests are
important.<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In a TCP/IP network many things can happen with IP
packets.
Firewalls,<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>NAT devices, Session Border Controllers and SIP Proxys are
in
the<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>signalling path and they will affect the
call.<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>And<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>What can I
do?<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>--------------<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Turn on SIP debug, try to understand the signalling that
happens<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>and see if you're missing the reply to the INVITE or
if the<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>ACK gets lost. When you know what happens, you've
taken the<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>first step to track down the problem. See the list above
and<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>investigate your
network.<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>For NAT and Firewall problems, there are many
documents<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>to help you. Start with reading sip.conf.sample that
is<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>part of your Asterisk
distribution.<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The SIP signalling standard, including
retransmissions<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>and timers for these, is well documented in the
IETF<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>RFC 3261.<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks,<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Matt G<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>:
http://www.voipphreak.ca<o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>: <a
href=3D"http://www.ratemydialplan.com">http://www.ratemydialplan.com</a><o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>: <a
href=3D"http://www.asterisk-jobs.com">http://www.asterisk-jobs.com</a>
</span><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>
<p class=3DMsoNormal><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt
0in 0in 0in'>
<p class=3DMsoNormal><b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] <b>On Behalf Of
</b>Todd Reese<br>
<b>Sent:</b> Friday, September 19, 2008 3:54 PM<br>
<b>To:</b> asterisk-users at lists.digium.com<br>
<b>Subject:</b> [asterisk-users] Dropping Phone
Calls<o:p></o:p></span></p>
</div>
<p class=3DMsoNormal><o:p> </o:p></p>
<div>
<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>Hi
All,<br>
<br>
<br>
I'm currently having trouble with dropped phone calls. The
following
error message is always in the log. This is a Grandstream GXP-2000
Firmware <a href=3D"http://1.1.6.16">1.1.6.16</a>
. The Asterisk box is
currently 1.4.22-rc5. The problem has been occurring on other versions
also.<br>
<br>
<br>
[Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission <a href=3D"mailto:8acaea6dc4c6e9b5 at
10.11.17.23">8acaea6dc4c6e9b5 at 10.11.17.23</a>
for seqno 50706 (Critical Response) -- See doc/sip-retransmit.txt.<br>
[Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1980 retrans_pkt: Hanging up call
<a
href=3D"mailto:8acaea6dc4c6e9b5 at 10.11.17.23">8acaea6dc4c6e9b5 at
10.11.17.23</a> -
no reply to our critical packet (see doc/sip-retransmit.txt).<br>
<br>
<br>
Any Ideas?<br>
<br>
<br>
Regards,<br>
<br>
Todd Reese<br>
<br>
<br>
<o:p></o:p></p>
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