Hi there, 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. People have reported issues with Polycom phones where the TLS thread crashes when the SDP is overloaded. http://community.polycom.com/t5/VoIP/IP550-Loses-Registration-as-soon-as-I-t... This is not just a Polycom issue, and doesn’t seem to be happening with other SIP servers. Many times, because the client doesn’t get a response, it sends the packets over and over and eventually times out. 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. Is someone willing to provide me with a SIP account where I can test TLS connections with SRTP enabled? An Echo test is enough. Thanks
On Mar 10, 2016, at 8:42 PM, Ken Rice <krice@freeswitch.org> wrote:
Are you sure you are sending this over TCP/TLS? This sounds like its really using UDP... TCP automatically does packet reassembly, UDP does not. The behavior you described sounds suspiciously like you are using UDP instead of TCP/TLS
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Emrah Sent: Thursday, March 10, 2016 1:37 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] SSL implementation in FreeSWITCH
Hi all, I’m writing to document where I’m at with my issues with FreeSWITCH and SSL / TLS and share my conclusions so far. 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.
First, I am running FreeSWITCH 1.6.6 on a Debian 8. Vars.xml shows sip_tls_version=tlsv1,tlsv1.1,tlsv1.2.
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.
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.
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.
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.
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.
I hope this message can be the starting point of a fruitful resolution process.
Thank you if you’ve read this up to here. Now hit reply and give me your 2 cents! :)
Best, Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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