[Freeswitch-users] need help of setting the proxy with kamailio server with freeswitch

sanjeev dubey sanjeevdb4 at gmail.com
Tue Jan 14 09:41:53 UTC 2020 

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 at 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 at lists.freeswitch.org> *Im
> Auftrag von *David Villasmil
> *Gesendet:* Freitag, 10. Januar 2020 13:59
> *An:* FreeSWITCH Users Help <freeswitch-users at 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 at 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 at lists.freeswitch.org> *Im
> Auftrag von *Alexander Haugg
> *Gesendet:* Donnerstag, 9. Januar 2020 08:17
> *An:* FreeSWITCH Users Help <freeswitch-users at 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 at MY_LOCAL_IP:4901
>
> BIND-URL
> sip:mod_sofia at 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 at 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 at lists.freeswitch.org
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> --
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.work at 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 at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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