<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="">Digging into my Yealink logs, I just wanted to show one interesting line that occurs right before the phone goes crazy on the TLS session.<div class=""><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class="">Mar 16 11:10:56 sua [574]: DLG <5+notice> [002] Message sent: (to dest=123.123.123.123:5061 len=1466)</div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class="">My little finger tells me the len=1466 is the size of the TLS record, or of its content, and it’s trying to squeeze it into one TCP segment.</div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class=""><br class=""></div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class="">I am just not sure what to do of this. Because it doesn’t seem to be a constant. I’ve had success with large packets I’m sure, but can’t say for sure in what network condition.</div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class=""><br class=""></div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class="">All I know for certain is that I have multiple phones and softphones in different network topologies. that when TLS and SRTP are used, calls randomly fail. If TCP is used with SRTP, it works fine. If TLS without SRTP is used, it also works consistently.</div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class=""><br class=""></div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class="">Emrah</div><div style="margin: 0px; font-size: 11px; line-height: normal; font-family: Menlo;" class=""><br class=""></div><div><blockquote type="cite" class=""><div class="">On Mar 13, 2016, at 9:24 PM, Emrah <<a href="mailto:lists@kavun.ch" class="">lists@kavun.ch</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><meta http-equiv="Content-Type" content="text/html charset=utf-8" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Scratch that, I need to confirm that this issue occurs when there is no NAT involved. <div class=""><br class=""></div><div class="">I’ll post here what happens.</div><div class=""><br class=""></div><div class="">Stan, I like your idea very much.</div><div class="">On Mar 13, 2016, at 7:58 PM, Emrah <<a href="mailto:lists@kavun.ch" class="">lists@kavun.ch</a>> wrote:<div class=""><blockquote type="cite" class=""><br class="Apple-interchange-newline"><div class=""><meta http-equiv="Content-Type" content="text/html charset=utf-8" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hi!<div class="">The server isn’t behind NAT, and I don’t know the make and model of all the routers that the affected clients use. And this problem also occurs when there isn’t NAT involved at all.</div><div class=""><br class=""></div><div class="">I will see to open a bug.</div><div class=""><br class=""></div><div class="">Best,</div><div class="">Emrah<br class=""><div class=""><blockquote type="cite" class=""><div class="">On Mar 11, 2016, at 11:12 PM, Brian West <<a href="mailto:brian@freeswitch.org" class="">brian@freeswitch.org</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">What kind of router is doing your nat? <div class=""><br class=""></div><div class="">Smells like someone needs 'ip virtual-reassembly'</div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Fri, Mar 11, 2016 at 4:09 PM, Ken Rice <span dir="ltr" class=""><<a href="mailto:krice@freeswitch.org" target="_blank" class="">krice@freeswitch.org</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">The particular message on the polycom boards only References FreeSWITCH 1.2... there have been significant changes from that point... without getting decoded packet captures theres really not much to see here.<br class="">
<br class="">
TCP handles the packet fragmentation as should TLS... this is transport protocol later stuff... this is not something FreeSWITCH would really be involved in at that point...<br class="">
<br class="">
The other thing that might prove useful is isolating this on an idle machine, with 1 phone enabling all the good SIP and libsofia debugging and getting it into a long along with the logs from the client so it can be compared to see whats going on...<br class="">
<br class="">
Further discussion of this should be taken to jira so that it can be tracked and captures and logging can be attached to the ticket<br class="">
<span class=""><br class="">
-----Original Message-----<br class="">
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" class="">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" class="">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Emrah<br class="">
</span><span class="">Sent: Friday, March 11, 2016 3:57 PM<br class="">
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" class="">freeswitch-users@lists.freeswitch.org</a>><br class="">
</span><div class=""><div class="h5">Subject: Re: [Freeswitch-users] SSL implementation in FreeSWITCH<br class="">
<br class="">
Hi there,<br class="">
I am using TLS over TCP for sure. I spent a significant amount of time inspecting this and UDP is only used for the RTP stream.<br class="">
People have reported issues with Polycom phones where the TLS thread crashes when the SDP is overloaded.<br class="">
<a href="http://community.polycom.com/t5/VoIP/IP550-Loses-Registration-as-soon-as-I-try-to-make-a-call/td-p/21372" rel="noreferrer" target="_blank" class="">http://community.polycom.com/t5/VoIP/IP550-Loses-Registration-as-soon-as-I-try-to-make-a-call/td-p/21372</a><br class="">
This is not just a Polycom issue, and doesn’t seem to be happening with other SIP servers.<br class="">
Many times, because the client doesn’t get a response, it sends the packets over and over and eventually times out.<br class="">
I doubt that the same SIP stack is used across Polycom, Yealink, Counterpath Bria, Blink Pro, etc… I will put together a PCAP of the session in TCP since I can’t share the private key of the SSL certificate.<br class="">
Is someone willing to provide me with a SIP account where I can test TLS connections with SRTP enabled? An Echo test is enough.<br class="">
<br class="">
Thanks<br class="">
> On Mar 10, 2016, at 8:42 PM, Ken Rice <<a href="mailto:krice@freeswitch.org" class="">krice@freeswitch.org</a>> wrote:<br class="">
><br class="">
> Are you sure you are sending this over TCP/TLS? This sounds like its<br class="">
> really using UDP... TCP automatically does packet reassembly, UDP does<br class="">
> not. The behavior you described sounds suspiciously like you are using<br class="">
> UDP instead of TCP/TLS<br class="">
><br class="">
> -----Original Message-----<br class="">
> From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" class="">freeswitch-users-bounces@lists.freeswitch.org</a><br class="">
> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" class="">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of<br class="">
> Emrah<br class="">
> Sent: Thursday, March 10, 2016 1:37 PM<br class="">
> To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" class="">freeswitch-users@lists.freeswitch.org</a>><br class="">
> Subject: [Freeswitch-users] SSL implementation in FreeSWITCH<br class="">
><br class="">
> Hi all,<br class="">
> I’m writing to document where I’m at with my issues with FreeSWITCH and SSL / TLS and share my conclusions so far.<br class="">
> I am hoping that this can give lieu to some further testing in different environments, and a proper fix if a bug is indeed confirmed.<br class="">
><br class="">
> First, I am running FreeSWITCH 1.6.6 on a Debian 8. Vars.xml shows sip_tls_version=tlsv1,tlsv1.1,tlsv1.2.<br class="">
><br class="">
> What I’ve observed is that in a sequence where client sens an invite to FS; FS responds with 407 proxy authorization required; client sends ack; Client sends the invite with the digest auth.<br class="">
><br class="">
> The last packet can easily exceed the max segment size of a TCP segment, typically if the SDP advertises a bunch of codecs, or if the client uses SRTP and the SAVP contains many crypto suites.<br class="">
><br class="">
> Now, when this occurs, the packets should be sent fragmented so they can fit in the MTU. It is then up to the receiving end to reassemble the segments and feed the complete packet to the application layer.<br class="">
><br class="">
> What I’ve noticed is that a packet that is too large is simply never received by FreeSWITCH. Since this is systematically the case with every software and hardware client I’ve used, I am drawn to think that the issue lies in the SSL implementation of FreeSWITCH.<br class="">
><br class="">
> In the event that for some reason my network or server OS configuration may be behind this, I would appreciate if someone would be willing to share some SIP credentials that can let me test TLS and SRTP. If getting to the bottom of this is of interest to any of you, I’d obviously be keen on handing out a couple of accounts.<br class="">
><br class="">
> I hope this message can be the starting point of a fruitful resolution process.<br class="">
><br class="">
> Thank you if you’ve read this up to here. Now hit reply and give me<br class="">
> your 2 cents! :)<br class="">
><br class="">
> Best,<br class="">
> Emrah<br class="">
> ______________________________________________________________________<br class="">
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