[asterisk-users] HELP! tls/srtp: sip_xmit error: returned -2
Da Rock
asterisk-users at herveybayaustralia.com.au
Mon Jun 13 21:19:20 CDT 2011
I know I've bumped this already now, but I do need to resolve this and
I've only been replying to myself.
I've tried another client now (Jitsi), which was the only one with
tls/srtp support that will run on freebsd, and it suffers the same problem.
I am very confused now as to why the only client that is demonstrated in
the docs is blink and is the only client to support a client
certificate. Is this the only way that this works- to have a server
_and_ a client certificate? Is this the source of the problem? Does this
mean asterisk is broken in this regard?
On 06/13/11 10:44, Da Rock wrote:
> I'm still no further advanced on this, but I think I have narrowed it
> down to tls. I have sip debug logs which shows that the server cannot
> contact the tls enabled phone at the same time this error crops up.
> The log says "calling <user>" and then the error.
>
> With TLS disabled, though, SRTP still doesn't work either though. I
> have no knowledge of how to move forward on this, so some pointers
> would be very much appreciated.
>
>
> On 06/07/11 12:11, Da Rock wrote:
>> I'm having trouble setting up tls/srtp secure communications on my
>> Asterisk server- I'm still rather new at working with Asterisk.
>>
>> I have enabled tls and encryption and I have csipsimple with tls
>> build on the phone. I'm currently only testing one phone with this
>> capability so far, and the rest still work in the current state.
>>
>> My logging looks like this with verbose turned up:
>>
>> [Jun 7 11:44:13] NOTICE[88483]: chan_sip.c:19842
>> handle_response_peerpoke: Peer '<user>' is now Reachable. (171ms /
>> 2000ms)
>> [Jun 7 11:46:17] NOTICE[88483]: chan_sip.c:25072 sip_poke_noanswer:
>> Peer '<user>' is now UNREACHABLE! Last qualify: 203
>> [Jun 7 11:46:29] NOTICE[88483]: chan_sip.c:19842
>> handle_response_peerpoke: Peer '<user>' is now Reachable. (1888ms /
>> 2000ms)
>>
>> When I call on this phone I get:
>>
>> [Jun 7 11:40:47] WARNING[88483]: chan_sip.c:3280 __sip_xmit:
>> sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2:
>> Invalid argument
>> [Jun 7 11:41:01] WARNING[88483]: chan_sip.c:3280 __sip_xmit:
>> sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2:
>> Invalid argument
>> [Jun 7 11:41:15] WARNING[88483]: chan_sip.c:3280 __sip_xmit:
>> sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2:
>> Invalid argument
>> [Jun 7 11:41:29] WARNING[88483]: chan_sip.c:3280 __sip_xmit:
>> sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2:
>> Invalid argument
>> -- Registered SIP '<user>' at 192.168.0.200:57805
>> [Jun 7 11:41:31] NOTICE[88483]: chan_sip.c:19842
>> handle_response_peerpoke: Peer '<user>' is now Reachable. (10ms /
>> 2000ms)
>>
>> When I call from another phone I get:
>>
>> [Jun 7 11:55:30] NOTICE[88483]: chan_sip.c:25072 sip_poke_noanswer:
>> Peer '<tls user>' is now UNREACHABLE! Last qualify: 13
>> -- SIP/<tls user>-00000024 is circuit-busy
>> == Everyone is busy/congested at this time (1:0/1/0)
>> -- Auto fallthrough, channel 'SIP/<user>-00000023' status is
>> 'CONGESTION'
>> [Jun 7 11:56:22] WARNING[88483]: chan_sip.c:3280 __sip_xmit:
>> sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:45931 returned -2:
>> Interrupted system call
>>
>> and eventually:
>>
>> [Jun 7 11:57:46] WARNING[88483]: chan_sip.c:3280 __sip_xmit:
>> sip_xmit of 0x2cefb000 (len 599) to 192.168.0.200:45931 returned -2:
>> Unknown error: 0
>>
>> I'm using my own CA setup for purposes beyond just this need, so I'm
>> using openssl commands directly and everything works elsewhere- so my
>> CA setup is fine (includes SAN).
>>
>> My config for tls/srtp looks like this (remember, the rest works very
>> happily):
>>
>> [global]
>> encryption = yes
>> tlsenable = yes
>> tlsbindaddr = 0.0.0.0
>> tlscertfile =
>> /path/to/asterisk/certificate/and/key/in/a/single/file
>> tlscafile = /path/to/CA/certificate
>> tlscipher = ALL
>> tlsclientmethod = tlsv1
>>
>> [tls user]
>> transport = tls
>>
>> Can someone give me any clues to what is happening? I've checked my
>> packet flow with tcpdump and wireshark as well, but I'm still left
>> mystified.
>>
>> Cheers
>>
>> --
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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