[SR-Users] NAT single interface problems with Lan/Wan

[Scott, Matt]

I can get wan communication or lan communication working, but not both.

Setting this, allows wan communication, but then my lan tries to talk with my wan ip.
Single Interface: listen=udp:<lan_ip>:5060 advertise <nat_wan_ip>:5060
I see my wan_ip in the via header, so I assume that's causing the issues for my lan.
Using add_contact_alias for lan, with the advertise in my listen command, results in  [nathelper.c:835]: no need to add alias param
I've tried handle_ruri_alias();, but this doesn't seem to have an affect either.


Setting this, allows lan communication to work fine, but wan has issues.
Single Interface: listen=udp:<lan_ip>:5060

Where is the happy median?


Was going to try path, but I read this is not the correct work-around.

Any clues or hints, are greatly appreciated.


U <carrier_wan_ip>:5060 -> <lan_ip>:5060
  INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9h
  G4bK04B0f1624ef34479ee3..From: <sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
  : <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
  VITE..Max-Forwards: 63..Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, appl
  ication/isup, application/dtmf, application/dtmf-relay,  multipart/mixed..Contact: <sip:+1720276
  3205@<carrier_wan_ip:5060>..P-Asserted-Identity: <sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported: 100
  rel..Content-Length:  305..Content-Disposition: session; handling=required..Content-Type: applic
  ation/sdp....v=0..o=Sonus_UAC 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media Capabilities..c=IN IP4
   4.55.10.130..t=0 0..m=audio 15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/800
  0..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-1
  5..a=sendrecv..a=maxptime:20..

U <lan_ip>:5060 -> <carrier_wan_ip>:5060
  SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP <carrier_wan_ip:5060;branch=z9h
  G4bK04B0f1624ef34479ee3..From: <sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
  : <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
  VITE..Server: kamailio (3.3.2 (x86_64/linux))..Content-Length: 0....

U <lan_ip>:5060 -> <asterisk_ip>:5060
  INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP <wan_ip>:5060;branch=
  z9hG4bK5d66.7cb4b7a1.0..Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3..Fro
  m: <sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To: <sip:+1<dialed_number>@192.168.
  9.130:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip>..CSeq: 27475 INVITE..Max-Forwards: 62..Allo
  w: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, application/isup, application/dt
  mf, application/dtmf-relay,  multipart/mixed..Contact: <sip:+1<caller_id>@<carrier_wan_ip:5060>..P-As
  serted-Identity: <sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported: 100rel..Content-Length:  305..C
  ontent-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UA
  C 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media Capabilities..c=IN IP4 4.55.10.130..t=0 0..m=audio
   15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=
  fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=maxptime:20
  ..


Asterisk box:

Retransmitting #4 (no NAT) to <wan_ip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <wan_ip>:5060;branch=z9hG4bK5d66.7cb4b7a1.0;received=192.168.9.130
Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3
From: <sip:+1<caller_id>@<carrier_wan_ip>:5060;isup-oli=0>;tag=gK0413c49b
To: <sip:+1<dialed_number>@<lan_ip>:5060>;tag=as1265f4d0
Call-ID: 184873373_18184370@<carrier_wan_ip>
CSeq: 27475 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:+1<dialed_number>@<asterisk_ip>>
Content-Type: application/sdp
Content-Length: 239





Matt Scott