[Spce-user] Remove local address from Peer INVITE Record-Route / Via Headers

[Nathaniel Pitcher]

Daniel,

Thanks for the reply. I actually tried that already. As you can see in the
invite I included the local address is now a 10. network instead of 127.

We still have the same issue where we receive the early media RTP and are
then disconnected about 30s in.

So there is no way to make the invite to the peer appear to originate from
the public ip?

I can see how having an internal address could be confusing once it is
routed to another network as it would not be able to route back.

Nathaniel Pitcher

-----
Sent from my mobile device. Please forgive any errors.

On Feb 18, 2017, at 4:19 AM, Daniel Grotti <dgrotti at sipwise.com> wrote:

Hi,
That's not possible, what you can do is to use a dummy interface and IP,
instead of localhost.
Please have a look at:
https://www.sipwise.org/doc/mr4.5.3/spce/ar01s04.html#_audiocodes_devices_workaround


Daniel


On Feb 18, 2017 3:58 AM, Nathaniel Pitcher <nathaniel at inmtn.net> wrote:

We are having a problem with some calls going to a peer.

We make a call and receive early media and then after about 30s the call is
terminated.

In speaking with our provider and giving them some pcaps, they seem to
think that their upstream provider does not like seeing “local” IPs in the
Record-Route or Via SIP headers.

Is there a way that we can only show the public IP when we send the invite
to a peer?





Below is the invite:


2017/02/18 01:48:59.218055 xxx.197.197.130:5060 -> xxx.115.69.144:5060
INVITE sip:12223334444 at xxx.115.69.144:5060;transport=udp SIP/2.0
Record-Route: <
sip:xxx.197.197.130;r2=on;lr=on;ftag=1DBB886B-58A7A80B0003501E-C6CEC700;ngcplb=yes
>
Record-Route: <
sip:10.111.111.111;r2=on;lr=on;ftag=1DBB886B-58A7A80B0003501E-C6CEC700;ngcplb=yes
>
Via: SIP/2.0/UDP
xxx.197.197.130;branch=z9hG4bK1dde.7f15d8fccb769b263f65fef36ad1e9be.0
Via: SIP/2.0/UDP 10.111.111.111:5080
;received=10.111.111.111;branch=z9hG4bK6ldVKae7;rport=5080
From: <sip:19518948809 at sip.server.net
>;tag=1DBB886B-58A7A80B0003501E-C6CEC700
To: <sip:12075552*17608021325 at xxx.115.69.144>
CSeq: 10 INVITE
Call-ID: 8b3190a6-3707-4bee-9322-97b7a4e31430_b2b-1
Max-Forwards: 69
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REGISTER
Supported: 100rel, replaces, norefersub
P-Asserted-Identity: <sip:15554443333 at sip.server.net>
Content-Type: application/sdp
Content-Length: 276
Contact: <
sip:ngcp-lb at xxx.197.197.130:5060;ngcpct=7369703a31302e3133382e302e3130353a35303830
>

v=0
o=- 899528559 899528559 IN IP4 172.16.123.123
s=Asterisk
c=IN IP4 xxx.9.52.19
t=0 0
m=audio 11590 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
a=maxptime:150
a=direction:both





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