[SR-Users] Issue with ACK behind NAT

[Alberto Sagredo]

Hi Alex

1.- Kamailio -> 172.26.101.50:8002 (Floating IP)

Asterisk -> 172.26.101.10:5080

2.-  Transmitting (no NAT) to 192.168.0.170:8002:

ACK sip:110 at IP_PUBLIC_IP:5066 SIP/2.0

Via: SIP/2.0/UDP 172.26.101.10:5080;branch=z9hG4bK0a330ae6

Route: <sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on>

Max-Forwards: 70

From: "asterisk" <sip:110 at 172.26.101.10:5080>;tag=as14d7523e

To: <sip:110 at 172.26.101.50:8002>;tag=1749303708

Contact: <sip:110 at 172.26.101.10:5080>

Call-ID: 7e723f1c64086e964df79e493350a2a4 at 172.26.101.10:5080

CSeq: 102 ACK

User-Agent: ast01

Content-Length: 0

With code i posted before i have now issue to answer calls from ast
(generated by asterisk)


200 From Phone arrives fine to Kamailio


<--- SIP read from UDP:172.26.101.50:8002 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.26.101.10:5080;rport=5080;branch=z9hG4bK125b3b98

Record-Route: <sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on>

From: "asterisk" <sip:110 at 172.26.101.10:5080>;tag=as14d7523e

To: <sip:110 at 172.26.101.50:8002>;tag=1749303708

Call-ID: 7e723f1c64086e964df79e493350a2a4 at 172.26.101.10:5080

CSeq: 102 INVITE

Contact: <sip:110 at PUBLIC.IP:5066>

Supported: replaces, path, timer, eventlist

User-Agent: Grandstream GXV3275 1.0.3.37

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE

Content-Type: application/sdp

Content-Length: 255


v=0

o=110 8003 8000 IN IP4 172.26.101.41

s=SIP Call

c=IN IP4 172.26.101.41

t=0 0

m=audio 8424 RTP/AVP 0 8 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sdp_proxied:yes

<------------->

I see 200 OK sent to Asterisk


Asterisk sent to me ACK but Kamailio seems to do not send to Phone behind
nat


3.- 192.168.0.170:8002


Thanks


2015-07-29 9:18 GMT+02:00 Alex Balashov <abalashov at evaristesys.com>:

> Alberto,
>
> 1. What are the literal (natively homed) IP addresses of Asterisk and
> Kamailio?
>
> 2. What is the Request Line (first line) of the ACK request being sent
> from Asterisk, i.e.
>
>    ACK sip:... SIP/2.0
>
> 3. To what IP and port is the ACK being sent by Asterisk?
>
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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>
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