[Freeswitch-users] Bridging between two rfc1918 networks

[Serge S. Yuriev]

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