Re: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working
Hi, if I set the parameter „<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> “ in the vars.xml and use this variable $${external_rtp_ip} in the SIP Profile, then the Ext-RTP-IP show the real external IP. I think, that’s OK. But I miss the srflx candidate in the sdp. I tried to experimental with the apply-candidate-acl (rfc1918.auto, any_v4.auto, wan_v4.auto), but nothing works. What’s wrong? Here is my sofia profile config: <profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua"> <aliases /> <domains> <domain name="h3k.sip.c4b" alias="false" parse="true" /> </domains> <settings> <param name="user-agent-string" value="XPhone Call Controller" /> <param name="debug" value="0" /> <param name="stun-enabled" value="true"/> <param name="stun-auto-disable" value="false"/> <param name="sip-trace" value="no" /> <param name="sip-capture" value="no" /> <param name="dialplan" value="XML" /> <param name="context" value="SBC2_SP_outbound" /> <param name="dtmf-type" value="rfc2833" /> <param name="dtmf-duration" value="100" /> <param name="rfc2833-pt" value="98" /> <param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" /> <param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" /> <param name="hold-music" value="local_stream://moh" /> <param name="rtp-timer-name" value="none" /> <param name="rtp-rewrite-timestamps" value="true" /> <param name="manage-presence" value="false" /> <param name="inbound-codec-negotiation" value="generous" /> <param name="inbound-late-negotiation" value="true" /> <param name="nonce-ttl" value="60" /> <param name="auth-calls" value="false" /> <param name="sip-port" value="4901" /> <param name="rtp-ip" value="MyLocalIP" /> <param name="sip-ip" value=" MyLocalIP " /> <param name="ext-rtp-ip" value="$${external_rtp_ip}" /> <param name="ext-sip-ip" value=" MyLocalIP " /> <param name="local-network-acl" value="SBC2_localnet.auto" /> <param name="apply-candidate-acl" value="rfc1918.auto" /> <param name="apply-candidate-acl" value="SBC2_localnet.auto" /> <param name="apply-candidate-acl" value="any_v4.auto" /> <param name="apply-candidate-acl" value="wan_v4.auto" /> <param name="apply-inbound-acl" value="SBC2_localnet.auto" /> <param name="rtp-timeout-sec" value="300" /> <param name="rtp-hold-timeout-sec" value="1800" /> <param name="enable-3pcc" value="true" /> <param name="inbound-use-callid-as-uuid" value="true" /> <param name="tls" value="false" /> <param name="tls-only" value="false" /> <param name="tls-bind-params" value="transport=tls" /> <param name="tls-sip-port" value="" /> <param name="tls-passphrase" value="" /> <param name="tls-verify-date" value="true" /> <param name="tls-verify-depth" value="2" /> <param name="tls-verify-in-subjects" value="" /> <param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" /> <param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" /> <param name="zrtp-passthru" value="true" /> <param name="inbound-reg-force-matching-username" value="true" /> <param name="auth-all-packets" value="false" /> <param name="wss-binding" value=":4447" /> <param name="caller-id-type" value="rpid" /> <param name="manual-redirect" value="true" /> <param name="rtcp-audio-interval-msec" value="5000" /> </settings> <gateways> <X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" /> </gateways> </profile> Von: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> Im Auftrag von Alexander Haugg Gesendet: Donnerstag, 9. Januar 2020 08:17 An: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Betreff: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working Hi, I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting. <param name="stun-enabled" value="true"/> <param name="stun-auto-disable" value="false"/> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org" /> The Freeswitch is started with the parameter –nonat In the pcap trace is no STUN request visible. I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request. If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”, then works it successfully. The status output of the profile is: ================================================================================================= Name SBC2_B2Bua Domain Name N/A Auto-NAT false DBName sofia_reg_SBC2_B2Bua Pres Hosts Dialplan XML Context SBC2_SP_outbound Challenge Realm auto_to RTP-IP MY_LOCAL_IP Ext-RTP-IP stun:stun.freeswitch.org SIP-IP MY_LOCAL_IP Ext-SIP-IP MY_LOCAL_IP URL sip:mod_sofia@MY_LOCAL_IP:4901 BIND-URL sip:mod_sofia@MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,PCMU,PCMA,VP8 CODECS OUT OPUS,PCMU,PCMA,VP8 TEL-EVENT 98 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 Thanks a lot!!! Alex
Does it work with a manually entered IP? On Fri, 10 Jan 2020 at 12:42, Alexander Haugg <Alexander.Haugg@c4b.de> wrote:
Hi,
if I set the parameter „<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> “ in the vars.xml and use this variable $${external_rtp_ip} in the SIP Profile, then the Ext-RTP-IP show the real external IP.
I think, that’s OK.
But I miss the srflx candidate in the sdp.
I tried to experimental with the apply-candidate-acl (rfc1918.auto, any_v4.auto, wan_v4.auto), but nothing works.
What’s wrong?
Here is my sofia profile config:
<profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi=" http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua">
<aliases />
<domains>
<domain name="h3k.sip.c4b" alias="false" parse="true" />
</domains>
<settings>
<param name="user-agent-string" value="XPhone Call Controller" />
<param name="debug" value="0" />
* <param name="stun-enabled" value="true"/>*
* <param name="stun-auto-disable" value="false"/>*
<param name="sip-trace" value="no" />
<param name="sip-capture" value="no" />
<param name="dialplan" value="XML" />
<param name="context" value="SBC2_SP_outbound" />
<param name="dtmf-type" value="rfc2833" />
<param name="dtmf-duration" value="100" />
<param name="rfc2833-pt" value="98" />
<param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
<param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
<param name="hold-music" value="local_stream://moh" />
<param name="rtp-timer-name" value="none" />
<param name="rtp-rewrite-timestamps" value="true" />
<param name="manage-presence" value="false" />
<param name="inbound-codec-negotiation" value="generous" />
<param name="inbound-late-negotiation" value="true" />
<param name="nonce-ttl" value="60" />
<param name="auth-calls" value="false" />
<param name="sip-port" value="4901" />
<param name="rtp-ip" value="MyLocalIP" />
<param name="sip-ip" value=" MyLocalIP " />
*<param name="ext-rtp-ip" value="$${external_rtp_ip}" />*
<param name="ext-sip-ip" value=" MyLocalIP " />
<param name="local-network-acl" value="SBC2_localnet.auto" />
* <param name="apply-candidate-acl" value="rfc1918.auto" />*
* <param name="apply-candidate-acl" value="SBC2_localnet.auto" />*
* <param name="apply-candidate-acl" value="any_v4.auto" />*
* <param name="apply-candidate-acl" value="wan_v4.auto" />*
<param name="apply-inbound-acl" value="SBC2_localnet.auto" />
<param name="rtp-timeout-sec" value="300" />
<param name="rtp-hold-timeout-sec" value="1800" />
<param name="enable-3pcc" value="true" />
<param name="inbound-use-callid-as-uuid" value="true" />
<param name="tls" value="false" />
<param name="tls-only" value="false" />
<param name="tls-bind-params" value="transport=tls" />
<param name="tls-sip-port" value="" />
<param name="tls-passphrase" value="" />
<param name="tls-verify-date" value="true" />
<param name="tls-verify-depth" value="2" />
<param name="tls-verify-in-subjects" value="" />
<param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" />
<param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" />
<param name="zrtp-passthru" value="true" />
<param name="inbound-reg-force-matching-username" value="true" />
<param name="auth-all-packets" value="false" />
<param name="wss-binding" value=":4447" />
<param name="caller-id-type" value="rpid" />
<param name="manual-redirect" value="true" />
<param name="rtcp-audio-interval-msec" value="5000" />
</settings>
<gateways>
<X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" />
</gateways>
</profile>
*Von:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> *Im Auftrag von *Alexander Haugg *Gesendet:* Donnerstag, 9. Januar 2020 08:17 *An:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Betreff:* [Freeswitch-users] WebRTC STUN request for a SIP profile is not working
Hi,
I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting.
<param name="stun-enabled" value="true"/>
<param name="stun-auto-disable" value="false"/>
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org" />
The Freeswitch is started with the parameter –nonat
In the pcap trace is no STUN request visible.
I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request.
If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”, then works it successfully.
The status output of the profile is:
=================================================================================================
Name SBC2_B2Bua
Domain Name N/A
Auto-NAT false
DBName sofia_reg_SBC2_B2Bua
Pres Hosts
Dialplan XML
Context SBC2_SP_outbound
Challenge Realm auto_to
RTP-IP MY_LOCAL_IP
Ext-RTP-IP stun:stun.freeswitch.org
SIP-IP MY_LOCAL_IP
Ext-SIP-IP MY_LOCAL_IP
URL sip:mod_sofia@MY_LOCAL_IP:4901
BIND-URL sip:mod_sofia@MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp
HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS IN OPUS,PCMU,PCMA,VP8
CODECS OUT OPUS,PCMU,PCMA,VP8
TEL-EVENT 98
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG true
PROXY-MEDIA false
ZRTP-PASSTHRU false
AGGRESSIVENAT false
CALLS-IN 0
FAILED-CALLS-IN 0
CALLS-OUT 0
FAILED-CALLS-OUT 0
REGISTRATIONS 0
Thanks a lot!!!
Alex
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
OK, I have the solution. 1. My think was to see the external candidate in the SDP. Freeswitch do that automatically, if the Client Registration from outside the local NW. 2. To force adding the external candidate, there is the channel var “include_external_ip=true”. For example -> originate {include_external_ip=true,media_webrtc=true}sofia/gateway/GW_SBC2_B2Bua/22100 &park 3. The srflx candidate will be set, if the internal port different from the external port (stun request). See the switch_core_media implementation for the candidate generation. My questions are solved (at the moment ;-). My biggest problem was to understand the server stack vs. client stack behaviour for WebRTC. Thanks a lot! Alex Von: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> Im Auftrag von David Villasmil Gesendet: Freitag, 10. Januar 2020 13:59 An: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Betreff: Re: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working Does it work with a manually entered IP? On Fri, 10 Jan 2020 at 12:42, Alexander Haugg <Alexander.Haugg@c4b.de<mailto:Alexander.Haugg@c4b.de>> wrote: Hi, if I set the parameter „<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org<http://stun.freeswitch.org>"/> “ in the vars.xml and use this variable $${external_rtp_ip} in the SIP Profile, then the Ext-RTP-IP show the real external IP. I think, that’s OK. But I miss the srflx candidate in the sdp. I tried to experimental with the apply-candidate-acl (rfc1918.auto, any_v4.auto, wan_v4.auto), but nothing works. What’s wrong? Here is my sofia profile config: <profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua"> <aliases /> <domains> <domain name="h3k.sip.c4b" alias="false" parse="true" /> </domains> <settings> <param name="user-agent-string" value="XPhone Call Controller" /> <param name="debug" value="0" /> <param name="stun-enabled" value="true"/> <param name="stun-auto-disable" value="false"/> <param name="sip-trace" value="no" /> <param name="sip-capture" value="no" /> <param name="dialplan" value="XML" /> <param name="context" value="SBC2_SP_outbound" /> <param name="dtmf-type" value="rfc2833" /> <param name="dtmf-duration" value="100" /> <param name="rfc2833-pt" value="98" /> <param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" /> <param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" /> <param name="hold-music" value="local_stream://moh" /> <param name="rtp-timer-name" value="none" /> <param name="rtp-rewrite-timestamps" value="true" /> <param name="manage-presence" value="false" /> <param name="inbound-codec-negotiation" value="generous" /> <param name="inbound-late-negotiation" value="true" /> <param name="nonce-ttl" value="60" /> <param name="auth-calls" value="false" /> <param name="sip-port" value="4901" /> <param name="rtp-ip" value="MyLocalIP" /> <param name="sip-ip" value=" MyLocalIP " /> <param name="ext-rtp-ip" value="$${external_rtp_ip}" /> <param name="ext-sip-ip" value=" MyLocalIP " /> <param name="local-network-acl" value="SBC2_localnet.auto" /> <param name="apply-candidate-acl" value="rfc1918.auto" /> <param name="apply-candidate-acl" value="SBC2_localnet.auto" /> <param name="apply-candidate-acl" value="any_v4.auto" /> <param name="apply-candidate-acl" value="wan_v4.auto" /> <param name="apply-inbound-acl" value="SBC2_localnet.auto" /> <param name="rtp-timeout-sec" value="300" /> <param name="rtp-hold-timeout-sec" value="1800" /> <param name="enable-3pcc" value="true" /> <param name="inbound-use-callid-as-uuid" value="true" /> <param name="tls" value="false" /> <param name="tls-only" value="false" /> <param name="tls-bind-params" value="transport=tls" /> <param name="tls-sip-port" value="" /> <param name="tls-passphrase" value="" /> <param name="tls-verify-date" value="true" /> <param name="tls-verify-depth" value="2" /> <param name="tls-verify-in-subjects" value="" /> <param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" /> <param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" /> <param name="zrtp-passthru" value="true" /> <param name="inbound-reg-force-matching-username" value="true" /> <param name="auth-all-packets" value="false" /> <param name="wss-binding" value=":4447" /> <param name="caller-id-type" value="rpid" /> <param name="manual-redirect" value="true" /> <param name="rtcp-audio-interval-msec" value="5000" /> </settings> <gateways> <X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" /> </gateways> </profile> Von: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> Im Auftrag von Alexander Haugg Gesendet: Donnerstag, 9. Januar 2020 08:17 An: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Betreff: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working Hi, I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting. <param name="stun-enabled" value="true"/> <param name="stun-auto-disable" value="false"/> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org<http://stun.freeswitch.org>" /> The Freeswitch is started with the parameter –nonat In the pcap trace is no STUN request visible. I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request. If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org<http://stun.freeswitch.org>”, then works it successfully. The status output of the profile is: ================================================================================================= Name SBC2_B2Bua Domain Name N/A Auto-NAT false DBName sofia_reg_SBC2_B2Bua Pres Hosts Dialplan XML Context SBC2_SP_outbound Challenge Realm auto_to RTP-IP MY_LOCAL_IP Ext-RTP-IP stun:stun.freeswitch.org<http://stun.freeswitch.org> SIP-IP MY_LOCAL_IP Ext-SIP-IP MY_LOCAL_IP URL sip:mod_sofia@MY_LOCAL_IP:4901 BIND-URL sip:mod_sofia@MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,PCMU,PCMA,VP8 CODECS OUT OPUS,PCMU,PCMA,VP8 TEL-EVENT 98 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 Thanks a lot!!! Alex _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work@gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337
Hi Team, Please help on the setting as proxy with kamailio sip server. Thanks On Fri, Jan 10, 2020, 7:33 PM Alexander Haugg <Alexander.Haugg@c4b.de> wrote:
OK,
I have the solution.
1. My think was to see the external candidate in the SDP. Freeswitch do that automatically, if the Client Registration from outside the local NW.
2. To force adding the external candidate, there is the channel var “include_external_ip=true”. For example -> originate {include_external_ip=true,media_webrtc=true}sofia/gateway/GW_SBC2_B2Bua/22100 &park
3. The srflx candidate will be set, if the internal port different from the external port (stun request). See the switch_core_media implementation for the candidate generation.
My questions are solved (at the moment ;-). My biggest problem was to understand the server stack vs. client stack behaviour for WebRTC.
Thanks a lot!
Alex
*Von:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> *Im Auftrag von *David Villasmil *Gesendet:* Freitag, 10. Januar 2020 13:59 *An:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Betreff:* Re: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working
Does it work with a manually entered IP?
On Fri, 10 Jan 2020 at 12:42, Alexander Haugg <Alexander.Haugg@c4b.de> wrote:
Hi,
if I set the parameter „<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> “ in the vars.xml and use this variable $${external_rtp_ip} in the SIP Profile, then the Ext-RTP-IP show the real external IP.
I think, that’s OK.
But I miss the srflx candidate in the sdp.
I tried to experimental with the apply-candidate-acl (rfc1918.auto, any_v4.auto, wan_v4.auto), but nothing works.
What’s wrong?
Here is my sofia profile config:
<profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi=" http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua">
<aliases />
<domains>
<domain name="h3k.sip.c4b" alias="false" parse="true" />
</domains>
<settings>
<param name="user-agent-string" value="XPhone Call Controller" />
<param name="debug" value="0" />
* <param name="stun-enabled" value="true"/>*
* <param name="stun-auto-disable" value="false"/>*
<param name="sip-trace" value="no" />
<param name="sip-capture" value="no" />
<param name="dialplan" value="XML" />
<param name="context" value="SBC2_SP_outbound" />
<param name="dtmf-type" value="rfc2833" />
<param name="dtmf-duration" value="100" />
<param name="rfc2833-pt" value="98" />
<param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
<param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
<param name="hold-music" value="local_stream://moh" />
<param name="rtp-timer-name" value="none" />
<param name="rtp-rewrite-timestamps" value="true" />
<param name="manage-presence" value="false" />
<param name="inbound-codec-negotiation" value="generous" />
<param name="inbound-late-negotiation" value="true" />
<param name="nonce-ttl" value="60" />
<param name="auth-calls" value="false" />
<param name="sip-port" value="4901" />
<param name="rtp-ip" value="MyLocalIP" />
<param name="sip-ip" value=" MyLocalIP " />
*<param name="ext-rtp-ip" value="$${external_rtp_ip}" />*
<param name="ext-sip-ip" value=" MyLocalIP " />
<param name="local-network-acl" value="SBC2_localnet.auto" />
* <param name="apply-candidate-acl" value="rfc1918.auto" />*
* <param name="apply-candidate-acl" value="SBC2_localnet.auto" />*
* <param name="apply-candidate-acl" value="any_v4.auto" />*
* <param name="apply-candidate-acl" value="wan_v4.auto" />*
<param name="apply-inbound-acl" value="SBC2_localnet.auto" />
<param name="rtp-timeout-sec" value="300" />
<param name="rtp-hold-timeout-sec" value="1800" />
<param name="enable-3pcc" value="true" />
<param name="inbound-use-callid-as-uuid" value="true" />
<param name="tls" value="false" />
<param name="tls-only" value="false" />
<param name="tls-bind-params" value="transport=tls" />
<param name="tls-sip-port" value="" />
<param name="tls-passphrase" value="" />
<param name="tls-verify-date" value="true" />
<param name="tls-verify-depth" value="2" />
<param name="tls-verify-in-subjects" value="" />
<param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" />
<param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" />
<param name="zrtp-passthru" value="true" />
<param name="inbound-reg-force-matching-username" value="true" />
<param name="auth-all-packets" value="false" />
<param name="wss-binding" value=":4447" />
<param name="caller-id-type" value="rpid" />
<param name="manual-redirect" value="true" />
<param name="rtcp-audio-interval-msec" value="5000" />
</settings>
<gateways>
<X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" />
</gateways>
</profile>
*Von:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> *Im Auftrag von *Alexander Haugg *Gesendet:* Donnerstag, 9. Januar 2020 08:17 *An:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Betreff:* [Freeswitch-users] WebRTC STUN request for a SIP profile is not working
Hi,
I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting.
<param name="stun-enabled" value="true"/>
<param name="stun-auto-disable" value="false"/>
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org" />
The Freeswitch is started with the parameter –nonat
In the pcap trace is no STUN request visible.
I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request.
If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”, then works it successfully.
The status output of the profile is:
=================================================================================================
Name SBC2_B2Bua
Domain Name N/A
Auto-NAT false
DBName sofia_reg_SBC2_B2Bua
Pres Hosts
Dialplan XML
Context SBC2_SP_outbound
Challenge Realm auto_to
RTP-IP MY_LOCAL_IP
Ext-RTP-IP stun:stun.freeswitch.org
SIP-IP MY_LOCAL_IP
Ext-SIP-IP MY_LOCAL_IP
URL sip:mod_sofia@MY_LOCAL_IP:4901
BIND-URL sip:mod_sofia@MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp
HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS IN OPUS,PCMU,PCMA,VP8
CODECS OUT OPUS,PCMU,PCMA,VP8
TEL-EVENT 98
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG true
PROXY-MEDIA false
ZRTP-PASSTHRU false
AGGRESSIVENAT false
CALLS-IN 0
FAILED-CALLS-IN 0
CALLS-OUT 0
FAILED-CALLS-OUT 0
REGISTRATIONS 0
Thanks a lot!!!
Alex
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
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FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
--
Regards,
David Villasmil
email: david.villasmil.work@gmail.com
phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
participants (3)
-
Alexander Haugg
-
David Villasmil
-
sanjeev dubey