[Freeswitch-users] How to get 200 Contact to use FQDN instead of IP

[E Chu]

Hi Michael,

The SIP client is an iOS app behind a NAT64 network interfacing to
Freeswitch which is on an IPV4 network.

In order for the 200 from FS to be properly acknowledged, the client needs
to send the ACK to the host in the Contact field of the 200.

However, if the Contact of the 200 contains the IPV4 address of FS, the
client isn't able to send the ACK to the IPV4 address from the NAT64
network.

So we're hoping that we can get FS to send its FQDN in the Contact field of
the 200.

Any suggestions appreciated.

Thanks!


---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Mon, 9 Apr 2018 12:17:21 -0400
> Subject: Re: [Freeswitch-users] How to get 200 Contact to use FQDN instead
> of IP
> why would you want to do this?
>
> On Apr 9, 2018, at 12:33 AM, E Chu <freeswitch-users at enlai.net> wrote:
>
> Hi all,
>
> Does anyone know how I can get FS to respond with a FQDN instead of an IP
> address in the contact field of the 200?
>
> The domain is set to FQDN in vars.xml:
>
> [hostnames and IPs changed for public list privacy]
>
> <X-PRE-PROCESS cmd=“set” data="domain=fs-dev1.sipserver.com"/>
>
> i.e. but this is what FS is sending in the 200:
>
> Contact: <sip:123123123 at 1.2.3.4:5060;transport=tcp>
>
> I’d like it to be:
>
> Contact: <sip:123123123 at fs-dev1.sipserver.com:5060;transport=tcp>
>
> This is the full 200 response from FS:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 192.168.0.1:52631
> ;rport=42810;branch=z9hG4bKPjwPkMUAMcou1XPnDrrWJH9GDwdlaPKXCF;alias;received=2.3.4.5
> From: "1020" <sip:1020 at fs-dev1.sipserver.com
> >;tag=h9ydSS-jO2dQ7xT8yxpVcLu-appl6Xtl
> To: <sip:123123123 at fs-dev1.sipserver.com>;tag=vXtN8yZgtHy0B
> Call-ID: Y3tWlOhNz3f9bt-uHO-LjFjh2RQwQh9l
> CSeq: 13167 INVITE
> Contact: <sip:123123123 at 1.2.3.4:5060;transport=tcp> //HOW TO GET THIS TO
> USE FQDN
> User-Agent: FreeSWITCH-mod_sofia/1.6.16~64bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Require: timer
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Session-Expires: 120;refresher=uac
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 248
> Remote-Party-ID: “123123123" <sip:123123123 at fs-dev1.sipserver.com
> >;party=calling;privacy=off;screen=no
>
> v=0
> o=FreeSWITCH 1522636701 1522636702 IN IP4 1.2.3.4
> s=FreeSWITCH
> c=IN IP4 1.2.3.4
> t=0 0
> m=audio 18966 RTP/AVP 0 96
> a=rtpmap:0 PCMU/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-16
> a=ptime:20
> a=rtcp:18967 IN IP4 1.2.3.4
>
>
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