[Freeswitch-users] Call is interrupted in 30 seconds, configuring NAT on Amazon EC2
Стас Тельнов
stasan89 at gmail.com
Fri Apr 8 18:34:21 MSD 2016
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?
>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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