No subject


Thu Jul 12 09:23:04 CDT 2007


 

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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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>From the =
&#8220;doc/sip-retransmit.txt&#8221;<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>&nbsp;</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>&nbsp;</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>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>- &quot;retrans_pkt: Hanging up call XX77yy&nbsp; - no =
reply to our
critical packet.&quot;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>- &quot;retrans_pkt: Cancelling retransmit of =
OPTIONs&quot;<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>&nbsp;</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>&nbsp;</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>&nbsp;</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>&nbsp;</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>&nbsp;</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>&nbsp;</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>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</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>&nbsp;</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>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; </span><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><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>&nbsp;</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>&nbsp;</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.&nbsp; The =
following
error message is always in the log.&nbsp; This is a Grandstream GXP-2000
Firmware <a href=3D"http://1.1.6.16">1.1.6.16</a> .&nbsp; The Asterisk =
box is
currently 1.4.22-rc5.&nbsp; 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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