<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hi Robert, thanks for the info, <span id="result_box" class="" tabindex="-1" lang="en">I upgraded my sipwise to the latest version 5.5.3 and the problem continued. Performing a debug on the side of our provider, he check that the ACK for some reason responded to the IP 127.0.0.1?<br class=""><span class="">finally we apply the WorkAround for the Audiocodes equipment and start receiving the ACK's correctly.</span></span><div class=""><span class="Apple-tab-span" style="white-space:pre"> </span><a href="https://www.sipwise.com/doc/mr5.5.3/spce/ar01s04.html#_audiocodes_devices_workaround" class="">https://www.sipwise.com/doc/mr5.5.3/spce/ar01s04.html#_audiocodes_devices_workaround</a><div class=""><br class=""></div><div class="">Thanks for your help</div><div class="">Jaime<br class=""><div class=""><div class="">
<br class=""><div><blockquote type="cite" class=""><div class="">El 23-02-2018, a las 7:01 a.m., Robert Cuaresma <<a href="mailto:rcuaresma@telcon.es" class="">rcuaresma@telcon.es</a>> escribió:</div><br class="Apple-interchange-newline"><div class="">
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Hi Jaime, <br class="">
<br class="">
I had the same problem few month ago... The problem was that the
carrier send me a very long call id. I solve the problem deleting
the name of the extensions on the PBX. This, reduce the SIP packet
length and may solve your problem (temporally). You can disable not
used codecs on pbx too.<br class="">
<br class="">
I hope can help you.<br class="">
<br class="">
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<title class=""></title>
<font size="-1" class=""><font face="Calibri" class="">Saludos,<br class="">
</font></font><span style="font-size:8.0pt;font-family:"Arial
Narrow",sans-serif;
color:#17365D;mso-ansi-language:CA;mso-fareast-language:ES" lang="CA" class=""><font size="-1" class=""><font face="Calibri" class="">Robert Cuaresma<br class="">
<br class="">
</font></font></span> </div>
<div class="moz-cite-prefix">El 21/02/2018 a las 17:39, Jaime
escribió:<br class="">
</div>
<blockquote type="cite" cite="mid:F8116241-50E4-4F36-8232-F4C50074F6DE@tnagroup.cl" class="">
<pre wrap="" class="">Hi All,
currently we use mr4.5.6 and works very well, but I have experienced a problem with incoming calls from a carrier.
The symptom is: the incoming calls from the Carrier to our Sipwise are cut at 30 sec.
I have seen that they are cut because in the SIP messages we have not received the ACK on the carrier side, which generates the cut at 30sec per timeout; and this is due to what I could say that the response sent to the Carrier: "SIP 200 OK” is fragmented because it is greater than 1500 bytes.
As I can see in the following wireshark:
4 5.539257 x.x.153.35 x.x.158.76 IPv4 1516 Fragmented IP protocol (proto=UDP 17, off=0, ID=9a96) [Reassembled in #5]
5 5.539274 x.x.153.35 x.x.158.76 SIP/SDP 197 Status: 200 OK |
Already raise this problem with our supplier, however I would like to ask if it is possible on our side (from the SipWise) to modify and reduce the size of the SIP packet response of the "SIP 200 OK", because I see that the answer and solution on the side of our carrier can take a long… (also my provider does not have SIP over TCP ;-(
Thanks in advance.
Jaime
In the SIP dialog I can see that multiple SIP 200 OK are sent but never receive an ACK:
=============================
|Time | x.x.158.76 |
| | | x.x.153.35 |
|0.000000 | INVITE SDP (g729 g71 |SIP INVITE From: <<a class="moz-txt-link-freetext" href="sip:955400000@x.x.158.76:5060">sip:955400000@x.x.158.76:5060</a> To:<a class="moz-txt-link-rfc2396E" href="sip:56228697777@x.x.153.35:5060Call-ID:7fabe7551f00-449edbc9-13c4-65014-9579e-6a8779fe-9579eCSeq:1||(5060)------------------"><sip:56228697777@x.x.153.35:5060 Call-ID:7fabe7551f00-449edbc9-13c4-65014-9579e-6a8779fe-9579e CSeq:1
| |(5060) ------------------></a> (5060) |
|0.002799 | 100 Trying| |SIP Status 100 Trying
| |(5060) <------------------ (5060) |
|0.128765 | 180 Ringing |SIP Status 180 Ringing
| |(5060) <------------------ (5060) |
|5.539274 | 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|6.018953 | 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|7.019163 | 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|11.017827| 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|15.018051| 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|19.017300| 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|23.018126| 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|27.018019| 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|31.018112| 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|35.017920| 200 OK SDP (g729 tel |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|37.524346| BYE | |SIP Request BYE CSeq:10
| |(5060) <------------------ (5060) |
|37.531809| 200 OK | |SIP Status 200 OK
| |(5060) ------------------> (5060) |
=============================
And the packet SIP 200 response, that is large than 1500 bytes.
=============================
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Message Header
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:127.0.0.1:5062;lr=on;ftag=7fabe8014538-449edbc9-13c4-65014-9579e-21b12904-9579e;did=7ad.8182;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=QVhMNGZ6N0p3S2lvUmEsOS4UXyMZFBFgNyQuHyITWVli"><sip:127.0.0.1:5062;lr=on;ftag=7fabe8014538-449edbc9-13c4-65014-9579e-21b12904-9579e;did=7ad.8182;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=QVhMNGZ6N0p3S2lvUmEsOS4UXyMZFBFgNyQuHyITWVli></a>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:127.0.0.1;r2=on;lr=on;ftag=7fabe8014538-449edbc9-13c4-65014-9579e-21b12904-9579e;ngcplb=yes;socket=udp:x.x.153.35:5060"><sip:127.0.0.1;r2=on;lr=on;ftag=7fabe8014538-449edbc9-13c4-65014-9579e-21b12904-9579e;ngcplb=yes;socket=udp:x.x.153.35:5060></a>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:x.x.153.35;r2=on;lr=on;ftag=7fabe8014538-449edbc9-13c4-65014-9579e-21b12904-9579e;ngcplb=yes;socket=udp:x.x.153.35:5060"><sip:x.x.153.35;r2=on;lr=on;ftag=7fabe8014538-449edbc9-13c4-65014-9579e-21b12904-9579e;ngcplb=yes;socket=udp:x.x.153.35:5060></a>
From: <a class="moz-txt-link-rfc2396E" href="sip:955400000@x.x.158.76:5060"><sip:955400000@x.x.158.76:5060></a>;tag=7fabe8014538-449edbc9-13c4-65014-9579e-21b12904-9579e
To: <a class="moz-txt-link-rfc2396E" href="sip:56228697777@x.x.153.35:5060"><sip:56228697777@x.x.153.35:5060></a>;tag=181C5B95-5A8502570005257D-46C9F700
Call-ID: 7fabe7551f00-449edbc9-13c4-65014-9579e-6a8779fe-9579e
CSeq: 1 INVITE
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:x.x.158.76:5060;transport=UDP;lr"><sip:x.x.158.76:5060;transport=UDP;lr></a>
Via: SIP/2.0/UDP x.x.158.76:5060;received=x.x.158.76;rport=5060;branch=z9hG4bK-5a850258-2244-b548b57
Via: SIP/2.0/UDP x.x.158.68:5060;branch=z9hG4bK-9579e-247e414d-51cc153d-7fabe8816328
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, MESSAGE, SUBSCRIBE
Supported: replaces, eventlist
User-Agent: Bria 3 release 3.5.5 stamp 71243
Content-Type: application/sdp
Content-Length: 275
Contact: <a class="moz-txt-link-rfc2396E" href="sip:ngcp-lb@x.x.153.35:5060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31"><sip:ngcp-lb@x.x.153.35:5060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31></a>
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 0 3 IN IP4 x.x.153.35
Session Name (s): Bria 3 release 3.5.5 stamp 71243
Connection Information (c): IN IP4 x.x.153.35
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 30362 RTP/AVP 18 101
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): fmtp:18 annexb=yes
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): direction:both
Media Attribute (a): sendrecv
Media Attribute (a): rtcp:30363
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</pre>
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