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@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@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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