[Freeswitch-users] Bridging between two rfc1918 networks

[Brian West]

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


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