[Freeswitch-users] Call is interrupted in 30 seconds, configuring NAT on Amazon EC2
Jurijs Ivolga
jurijs.ivolga at gmail.com
Fri Apr 8 18:37:44 MSD 2016
Hi,
I would recommend you to capture SIP packets during call on Freeswitch
server and send it here, I will take a look on it.
With kind regards,
Jurijs
On Fri, Apr 8, 2016 at 5:34 PM, Стас Тельнов <stasan89 at gmail.com> wrote:
> I already tried disabling timers, does not work.
>
> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>
>> Try disabling session timers in the sip profile. I think that line is
>> commented out by default, so uncomment it.
>>
>> <param name="enable-timer" value="false"/>
>>
>> On Fri, Apr 8, 2016 at 6:59 AM, Стас Тельнов <stasan89 at gmail.com> wrote:
>>
>>> Hello.
>>>
>>> When using a call or conference through sip — freeswitch with external
>>> provider there is a problem – the call is interrupted in 30 seconds. Though
>>> the sound goes all right.
>>> I think that it caused by the NAT settings for freeswitch, but I don't
>>> understand how to adjust it correctly.
>>> At start of freeswitch I see the following mistakes in the tracking data:
>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP
>>> [general error]
>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
>>> devices detected!
>>>
>>> Despite of this mistake, conference communication between two internal
>>> users works normally. The problem arises at a call through external
>>> provider.
>>>
>>> We have the following architecture:
>>> In a cloud of Amazon EC2 there are 2 servers – opensips and freeswitch,
>>> both for NAT for external clients, but have an opportunity to work with
>>> each other directly.
>>> opensips has the internal address 172.31.0.169 and external 52. *.*.177
>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>> *.*.198
>>>
>>> In fact, freeswitch acts only for conferences, and is ready for use of a
>>> remote DB on opensips.
>>> The auto-nat settings by default didn't work. The problem is in the
>>> external profile settings as far as I understand.
>>>
>>> I have filled and created the following configuration:
>>> vars.xml
>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto”/>
>>> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198”/> <!—
>>> public freeswitch ip —>
>>> <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198”/> <!—
>>> public freeswitch ip —>
>>> <!-- External SIP Profile -->
>>> <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>> <X-PRE-PROCESS cmd="set"
>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>
>>> sip_profile/external.xml
>>> <param name="rtp-ip" value="$${local_ip_v4}"/>
>>> <param name="sip-ip" value="$${local_ip_v4}"/>
>>>
>>> <param name="ext-rtp-ip" value=“52.*.*.198”/> <!— public freeswitch
>>> ip —>
>>> <param name="ext-sip-ip" value=“52.*.*.198”/> <!— public freeswitch
>>> ip —>
>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>> (that would be logical), but in that case conferences didn't work at all
>>> and errors below appeared:
>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>> Also I tried to put such configuration:
>>> <param name="rtp-ip" value="auto"/>
>>> <param name="sip-ip" value="52.*.*.198”/>
>>> but it also hasn't helped to solve the problem.
>>>
>>> autoload_configs/switch.conf.xml
>>> <param name="rtp-start-port" value="16384"/>
>>> <param name="rtp-end-port" value="32768"/>
>>>
>>> "sofia status" looks as follows:
>>> Name Type
>>> Data State
>>>
>>> =================================================================================================
>>> 172.31.22.124 alias
>>> internal ALIASED
>>> external profile sip:mod_sofia at 52.*.*.198:5060
>>> RUNNING (0)
>>> external profile sip:mod_sofia at 52.*.*.198:5061
>>> RUNNING (0) (TLS)
>>> external::*********.com gateway sip:USER@*********.com
>>> REGED
>>> internal profile sip:mod_sofia at 52.*.*.198:5080
>>> RUNNING (0)
>>> internal profile sip:mod_sofia at 52.*.*.198:5081
>>> RUNNING (0) (TLS)
>>>
>>> =================================================================================================
>>> 2 profiles 1 alias
>>>
>>> "sofia status profile external" looks as follows:
>>>
>>> =================================================================================================
>>> Name external
>>> Domain Name N/A
>>> Auto-NAT false
>>> DBName sofia_reg_external
>>> Pres Hosts
>>> Dialplan XML
>>> Context public
>>> Challenge Realm auto_to
>>> RTP-IP 172.31.22.124
>>> Ext-RTP-IP 52.*.*.198
>>> SIP-IP 172.31.22.124
>>> Ext-SIP-IP 52.*.*.198
>>> URL sip:mod_sofia at 52.*.*.198:5060
>>> BIND-URL sip:mod_sofia at 52.
>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>> TLS-URL sip:mod_sofia at 52.*.*.198:5061
>>> TLS-BIND-URL sips:mod_sofia at 52.
>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>> HOLD-MUSIC local_stream://moh
>>> OUTBOUND-PROXY N/A
>>> CODECS IN PCMA
>>> CODECS OUT PCMA
>>> TEL-EVENT 101
>>> DTMF-MODE rfc2833
>>> CNG 13
>>> SESSION-TO 0
>>> MAX-DIALOG 0
>>> NOMEDIA false
>>> LATE-NEG true
>>> PROXY-MEDIA false
>>> ZRTP-PASSTHRU true
>>> AGGRESSIVENAT false
>>> CALLS-IN 0
>>> FAILED-CALLS-IN 0
>>> CALLS-OUT 0
>>> FAILED-CALLS-OUT 0
>>> REGISTRATIONS 0
>>>
>>>
>>>
>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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