On Fri, Apr 8, 2016 at 6:59 AM, Стас Тельнов <stasan89@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 —>

<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@52.*.*.198:5060    RUNNING (0)
                 external    profile               sip:mod_sofia@52.*.*.198:5061    RUNNING (0) (TLS)
external::*********.com    gateway                      sip:USER@*********.com    REGED
                 internal    profile               sip:mod_sofia@52.*.*.198:5080    RUNNING (0)
                 internal    profile               sip:mod_sofia@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@52.*.*.198:5060
BIND-URL             sip:mod_sofia@52.*.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
TLS-URL              sip:mod_sofia@52.*.*.198:5061
TLS-BIND-URL         sips:mod_sofia@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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