[Freeswitch-users] Bridging between two rfc1918 networks

Anthony Minessale anthony.minessale at gmail.com
Sat Oct 8 05:23:03 MSD 2016 

Too terse.

You probably need to produce full traces on pastebin with the full debug to
get any idea.


On Fri, Oct 7, 2016 at 6:13 PM, Serge Yuriev <me at nevian.org> wrote:

> As mentioned before I tried to play with local-network-acl but no joy.
> Maybe it’s just not right? On which profile I should tune?
>
>    <list name="lan" default="deny">
>       <node type="deny" cidr="172.17.2.3/32"/>
>       <node type="deny" cidr="172.17.2.4/32"/>
>       <node type="allow" cidr="192.168.0.0/16"/>
>       <node type="allow" cidr="10.0.0.0/8"/>
>       <node type="allow" cidr="172.16.0.0/12"/>
>     </list>
>
> On both profiles I have like this
> Int
>    <param name="rtp-ip" value="$${inside_bind_ipv4}"/>
>    <param name="sip-ip" value="$${inside_bind_ipv4}"/>
>    <param name="ext-rtp-ip" value="$${inside_bind_ipv4}"/>
>    <param name="ext-sip-ip" value="$${inside_bind_ipv4}"/>
>
> Ext
>     <param name="rtp-ip" value="$${outside_bind_ipv4}"/>
>     <param name="sip-ip" value="$${outside_bind_ipv4}"/>
>     <param name="ext-rtp-ip" value="$${outside_bind_ipv4}"/>
>     <param name="ext-sip-ip" value="$${outside_bind_ipv4}"/>
>
> On 8 Oct  2016, at 00:48, Brian West <brian at freeswitch.org> wrote:
>
> you have to fix your local-network-acl in each system probably to do the
> right thing, do you have the ext-rtp-ip set with the automat: prefix?
>
> On Fri, Oct 7, 2016 at 1:23 PM, Serge S. Yuriev <me at nevian.org> wrote:
>
>> Hello,
>>
>> Two SIP profiles:
>> External 83.хх
>> Internal 10.23.154.0/24
>>
>> Via external we are receiving/send calls from/to 172.17.2.0/29
>> For some reason if we call outside FS sends unmodified addresses in SDP.
>> So we have unroutable address in SDP and one-way audio. If call flows
>> ext to int all working correct.
>> Tried local-network-acl on inside (10.хх) with excluded 172.хх,
>> apply-nat-acl with included 172.xx on either int and ext. Nothing helps :(
>>
>> "Bad one" SDP - from internal to external
>> send 960 bytes to udp/[10.23.154.63]:6060 at 18:16:22.226984:
>>     ------------------------------------------------------------
>> ------------
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP 10.23.154.63:6060;branch=z9hG4bKe433fa68b81
>>     From: "IT, Юрьев Сергей"
>> <sip:12550 at 10.23.154.63>;tag=195594~27154efa-6325-45a2-9e47-
>> 67e5d9302ebc-237816120
>>     To: <sip:62987%236546 at 10.23.154.100>;tag=66NUXXHvB6HBp
>>     Call-ID: 86c80-7f71bc46-c44e-3f40000a at 10.23.154.63
>>     CSeq: 101 INVITE
>>     Contact: <sip:mod_sofia at 10.23.154.100:6060>
>>     User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160707T165535Z~be13536ac9~64bit
>>     Accept: application/sdp
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> PRACK, NOTIFY
>>     Require: timer
>>     Supported: precondition, 100rel, timer, path, replaces
>>     Allow-Events: talk, hold, conference, refer
>>     Session-Expires: 1800;refresher=uac
>>     Content-Type: application/sdp
>>     Content-Disposition: session
>>     Content-Length: 180
>>
>>     v=0
>>     o=- 1475853382 2 IN IP4 172.17.2.3
>>     s=-
>>  >>   c=IN IP4 172.17.2.4
>>     b=AS:64
>>     t=0 0
>>     m=audio 3040 RTP/AVP 8 101
>>     a=rtpmap:8 PCMA/8000
>>     a=rtpmap:101 telephone-event/8000
>>     a=ptime:20
>>
>>
>> And a good one - external to internal
>> send 1162 bytes to udp/[10.23.154.65]:5060 at 12:34:15.132027:
>>     ------------------------------------------------------------
>> ------------
>>     INVITE sip:12550 at 10.23.154.65 SIP/2.0
>>     Via: SIP/2.0/UDP 10.23.154.100:6060;rport;branch=z9hG4bKUXyFjDmg8rtmB
>>     Max-Forwards: 69
>>     From: "Абонент"
>> <sip:$(caller_id_number)@10.23.154.100>;tag=1agg8aZ7FUUBK
>>     To: <sip:12550 at 10.23.154.65>
>>     Call-ID: d8367628-0fc1-4325-998f-3f32f9d3a05b
>>     CSeq: 97580363 INVITE
>>     Contact: <sip:gw+cucm-65 at 10.23.154.100:6060;transport=udp;gw=cucm-65>
>>     User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160707T165535Z~be13536ac9~64bit
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> PRACK, NOTIFY
>>     Supported: precondition, 100rel, timer, path, replaces
>>     Allow-Events: talk, hold, conference, refer
>>     Content-Type: application/sdp
>>     Content-Disposition: session
>>     Content-Length: 268
>>     X-FS-Support: update_display,send_info
>>     Remote-Party-ID: "Абонент"
>> <sip:$(caller_id_number)@10.23.154.100>;party=calling;screen
>> =yes;privacy=off
>>
>>     v=0
>>     o=FreeSWITCH 1475804423 1475804424 IN IP4 10.23.154.100
>>     s=FreeSWITCH
>>  >>   c=IN IP4 10.23.154.100
>>     t=0 0
>>     m=audio 28432 RTP/AVP 8 18 101 13
>>     a=rtpmap:8 PCMA/8000
>>     a=rtpmap:18 G729/8000
>>     a=rtpmap:101 telephone-event/8000
>>     a=fmtp:101 0-16
>>     a=rtpmap:13 CN/8000
>>     a=ptime:20
>>
>> --
>> Serge S. Yuriev
>> Senior VoIP engineer
>>
>>
>> _________________________________________________________________________
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>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
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>
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>
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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
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>
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> http://www.freeswitch.org
>
>
> --
> Serge S. Yuriev
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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