[Freeswitch-users] SRTP breaks my TLS session
Emrah
lists at kavun.ch
Tue Mar 1 14:45:05 MSK 2016
Hello all,
Pinging the list in search of brilliant minds for more recommendations on how to solve this overwhelmingly frustrating issue.
I just love FS and all it lets me accomplish. But this bug and the unpredictability and inability to effectively troubleshoot just leaves a bad impression, accentuated by the fact that no one seems to care about it. I call it a bug because I do not experience this issue with other SIP servers.
Who is available to exchange SIP credentials in order to cross-test this? Especially if you believe you are immunized!
Best,
Emrah
> On Feb 28, 2016, at 7:33 PM, Emrah <lists at kavun.ch> wrote:
>
> Hello there,
> I can confirm that a PCAP gives me a bunch of TLS packets.
> How do you suggest debugging this?
>
> Thanks!
>> On Feb 26, 2016, at 10:36 AM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
>>
>> Thanks for this.
>>
>> This isn’t just a yealink thing. I’ve encountered sporadic issues with soft phones and other desk phones as well.
>>
>> I didn’t use the PCap capture feature because I had assumed it would give me a bunch of TLS packets. I’ll test that and revert back.
>>
>> How can we explain that I have more calls failing if I register multiple accounts?
>>
>> Emrah
>>
>>> On Feb 26, 2016, at 6:37 AM, Mario G <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>>>
>>> You may want to run a pcap trace on the Yealink. It’s under settings->Configuration. Start/test/export.
>>>
>>>> On Feb 25, 2016, at 3:35 PM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>>>
>>>> Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections.
>>>>
>>>> On Thu, Feb 25, 2016 at 5:13 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
>>>> Hello list,
>>>> I thought I had solved this issue by reducing my codec list to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size.
>>>> Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS.
>>>> Incoming calls work reliably with or without SRTP.
>>>>
>>>> How do you suggest debugging this?
>>>> I tried setting up a fresh instance of FS but the issue persists.
>>>> Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device.
>>>>
>>>> Any suggestion is welcome. Have you experienced this?
>>>>
>>>> I’m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95.
>>>>
>>>> E
>>>>
>>>> _________________________________________________________________________
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>>>>
>>>>
>>>> --
>>>> Brian West
>>>> brian at freeswitch.org <mailto:brian at freeswitch.org>
>>>>
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>>>>
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>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>
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>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
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>>>
>>> _________________________________________________________________________
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>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>
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>>>
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>>
>
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