asterisk users - Oct 2016 - Registered successfully, but after a minute or so no SIP messages anymore

Hi all,
I have an issue with asterisk 13 and pjsip. I guess it is somehow 
Firewall related, but I'm unsure.

A registration to Sipgate is established successfully:


<Registration/ServerURI..............................>
<Auth..........>
<Status.......>
=========================================================================================
  pjsip_sipgate/sip:sipgate.de:5060 pjsip_sipgate     Registered


Calling the registered number is even successfully shown in asterisk (it 
is a freepbx installation).
But when doing a second call the number is busy ("provider" busy, I 
don't see anything in asterisk verbose mode).
Sending a pjsip unregister results in the following messages:

[2016-10-15 10:03:22] WARNING[10162]: 
res_pjsip_outbound_registration.c:761 schedule_retry: No response 
received from 'sip:sipgate.de:5060' on registration attempt to 
'sip:2636146e0 at sipgate.de:5060', retrying in '60'
     -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now 
Reachable.  RTT: 434.393 msec
   == Endpoint pjsip_sipgate is now Reachable

so it is somewhat clear, why i get a busy, because the endpoint is not 
reachable. But WHY is the endpoint not reachable?

Regarding the architecture: I have two routers cascaded, that is 
unfortunately necessary. On the first router (vDSL-access router) I have 
forwarded nearly everything to the second router (Bintec rj 353), where 
a port forwarding for relevant ports (sip and pjsip (udp and tcp), rtp 
(udp)) is configured. IF a call goes through, nearly everything is 
working (audio only incoming, but that is another issue).

STUN is configured. FreePBX Firewall is disabled.

Kind regards,
andre


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Very interesting: I have another provider configured, that was not 
reachable as well. I disabled the STUN-server (external STUN server), 
and now the second registration works fine, BUT with the same "error" 
messages (unreachable etc) as the other provider. But in contrast the 
number is always reachable!!!

Is there any explanation for this? I just want to understand... ;-) ... 
and solve it.

regards,
andre

Am 15.10.2016 um 10:11 schrieb Andre Gronwald:
>
> [2016-10-15 10:03:22] WARNING[10162]: 
> res_pjsip_outbound_registration.c:761 schedule_retry: No response 
> received from 'sip:sipgate.de:5060' on registration attempt to 
> 'sip:2636146e0 at sipgate.de:5060', retrying in '60'
>     -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now 
> Reachable.  RTT: 434.393 msec
>   == Endpoint pjsip_sipgate is now Reachable
>
> so it is somewhat clear, why i get a busy, because the endpoint is not 
> reachable. But WHY is the endpoint not reachable?
>
> Regarding the architecture: I have two routers cascaded, that is 
> unfortunately necessary. On the first router (vDSL-access router) I 
> have forwarded nearly everything to the second router (Bintec rj 353), 
> where a port forwarding for relevant ports (sip and pjsip (udp and 
> tcp), rtp (udp)) is configured. IF a call goes through, nearly 
> everything is working (audio only incoming, but that is another issue).
>
> STUN is configured. FreePBX Firewall is disabled.
>
> Kind regards,
> andre
>
>
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All other things aside, this stands out immediately:

RTT: 434.393 msec

That's almost half a second round trip for a packet. I'm amazed
anything works at all. For SIP connections, mine are usually about
26ms max, anything above about 35 is bad. Looks like a serious config
issue.

Try pinging and see what you get - my ping times to sipgate.de from
the UK are Best:13.6ms Worst 13.8ms across 100 pings.

I could be wrong, but I'd be surprised if that wasn't causing
problems, at least with audio.


On 15 October 2016 at 09:11, Andre Gronwald <andregronwald78 at gmail.com>
wrote:
> Hi all,
> I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall
> related, but I'm unsure.
>
> A registration to Sipgate is established successfully:
>
>
> <Registration/ServerURI..............................> 
<Auth..........>
> <Status.......>
>
=========================================================================================>
>  pjsip_sipgate/sip:sipgate.de:5060                       pjsip_sipgate
> Registered
>
>
> Calling the registered number is even successfully shown in asterisk (it is
> a freepbx installation).
> But when doing a second call the number is busy ("provider" busy,
I don't
> see anything in asterisk verbose mode).
> Sending a pjsip unregister results in the following messages:
>
> [2016-10-15 10:03:22] WARNING[10162]: res_pjsip_outbound_registration.c:761
> schedule_retry: No response received from 'sip:sipgate.de:5060' on
> registration attempt to 'sip:2636146e0 at sipgate.de:5060',
retrying in '60'
>     -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now
Reachable.
> RTT: 434.393 msec
>   == Endpoint pjsip_sipgate is now Reachable
>
> so it is somewhat clear, why i get a busy, because the endpoint is not
> reachable. But WHY is the endpoint not reachable?
>
> Regarding the architecture: I have two routers cascaded, that is
> unfortunately necessary. On the first router (vDSL-access router) I have
> forwarded nearly everything to the second router (Bintec rj 353), where a
> port forwarding for relevant ports (sip and pjsip (udp and tcp), rtp (udp))
> is configured. IF a call goes through, nearly everything is working (audio
> only incoming, but that is another issue).
>
> STUN is configured. FreePBX Firewall is disabled.
>
> Kind regards,
> andre
>
>
>
> --
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ping times are fine as well:

[root at freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57 time=46.8 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=5 ttl=57 time=47.1 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=6 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=7 ttl=57 time=47.1 ms
^C
--- sipgate.de ping statistics ---
7 packets transmitted, 7 received, 0% packet loss, time 6360ms
rtt min/avg/max/mdev = 46.406/46.809/47.191/0.393 ms
[root at freepbx asterisk]#


this high RTT appears only sometimes.  After removing STUN-server it 
looks better, did two test calls right now, both gone through 
immediately. At the end of the second test call I see:

     -- Executing [s at app-announcement-1:5] 
Playback("PJSIP/pjsip_sipgate-00000003", 
"custom/araz01&custom/07-polly,noanswer") in new stack
     -- <PJSIP/pjsip_sipgate-00000003> Playing
'custom/araz01.alaw'
(language 'en')
     -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now 
Reachable.  RTT: 493.094 msec
   == Endpoint pjsip_sipgate is now Reachable
     -- <PJSIP/pjsip_sipgate-00000003> Playing
'custom/07-polly.slin'
(language 'en')
     -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now 
Unreachable.  RTT: 0.000 msec
*  == Endpoint pjsip_sipgate is now Unreachable*


Why do I have that loss of registrations?

here my pjsip config for sipgate.de:

freepbx*CLI> pjsip show registration pjsip_sipgate

  <Registration/ServerURI..............................> 
<Auth..........>  <Status.......>
=========================================================================================
  pjsip_sipgate/sip:sipgate.de:5060 pjsip_sipgate     Registered

  ParameterName            : ParameterValue
  ======================================================= 
auth_rejection_permanent : true
  client_uri               : sip:2636146e0 at sipgate.de:5060
  contact_user             : 2636146e0
  endpoint                 :
  expiration               : 600
  fatal_retry_interval     : 0
  forbidden_retry_interval : 0
  line                     : false
  max_retries              : 10
  outbound_auth            : pjsip_sipgate
  outbound_proxy           :
  retry_interval           : 60
  server_uri               : sip:sipgate.de:5060
  support_path             : false
  transport                : 0.0.0.0-udp

Remind: Endpoint is currently unreachable, but asterisk shows 
"Registered". Test call fails at this moment.


regards,
andre
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