[Freeswitch-users] sip trunk question: why call through external profile is challenged?

Wed Jan 13 06:10:02 PST 2010

Hi all!

I'm brand new to FreeSwitch, but have some experience with SipX.

Our company is evaluating FS.

For test purposes I set up FS on a virtual machine (vmware esxi). I use
CentOS.

The FS version I use is 1.0.5-20100110-0400.

I have a question regarding sip trunk between FS and SipX.

I created the following GW in external profile: 

[freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'

<include>

  <gateway name="sipx4.lab.nstel.ru">

  <param name="username" value="zxcv"/>

  <param name="password" value="2007"/>

  <param name="register" value="false"/>

  </gateway>

</include>

External.xml file is not modified after installation.

I see this gateway via fs_cli:

freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru

============================================================================
=====================

Name            sipx4.lab.nstel.ru

Profile         external

Scheme          Digest

Realm           sipx4.lab.nstel.ru

Username        zxcv

Password        yes

>From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>

Contact
<sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nste
l.ru>

Exten           zxcv

To              sip:zxcv at sipx4.lab.nstel.ru

Proxy           sip:sipx4.lab.nstel.ru

Context         public

Expires         3600

Freq            3600

Ping            0

PingFreq        0

PingState       0/0/0

State           NOREG

Status          UP

CallsIN         0

CallsOUT        0

============================================================================
=====================

I created new FS extension 2853. I registers (xlite) and I can call it.

Now I want to call FS user from sipx.

>From the sipx side one can configure link to FS differently. There are two
options:

1. Call FS directly trough sipxproxy (the core part of sipx works as sip
proxy, not as b2bua)

2. Call trough embedded b2bua, named sipxbridge.

When a call is going trough sipxbridge, it is successfully landed at FS
extension 2853.

But when a call is going from the sipxproxy, it is challenged with Status
407 "proxy authentication required', and then call fails.

(I'm not sure if sipx should handle this challenge, but this is separate
question for the sipx forum).

At the default log level I see nothing in the freeswitch.log.

So the question is why one call through external profile is being challenged
while the other is not?

I suspect that the reason is in the difference in the two Invite messages:

Here is challenged Invite:

INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0

Record-Route:
<sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxN
zBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60
%213ba6d85f6946c4c6001bee1d3b54474f>

From:
"testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d
6e07e-2f6dec

To: <sip:2853 at lab.nstel.ru>

Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru

Cseq: 2 INVITE

Via: SIP/2.0/UDP
172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847

Via: SIP/2.0/TCP
172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac
7fd729330fd563f83cacd941311e75;id=22260-1

Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784

Max-Forwards: 18

Supported: replaces

User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634

Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>

Proxy-Authorization: Digest
username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb8
4b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044
ccaa",algorithm=MD5

Content-Type: application/sdp

Content-Length: 301

Date: Wed, 13 Jan 2010 11:47:22 GMT

Expires: 60

X-Sipx-Handled: X172.23.12.104-81.211.30.104

v=0

o=LGEIPP 16246 16247 IN IP4 172.23.12.233

s=SIP Call

c=IN IP4 172.23.12.233

t=0 0

m=audio 23008 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

And here is successful Invite:

INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0

Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0

CSeq: 1 INVITE

From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961

To: <sip:2853 at fs.lab.nstel.ru;user=phone>

Via: SIP/2.0/UDP
172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534

Max-Forwards: 70

User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)

P-Asserted-Identity: <sip:1004 at 172.23.12.104>

Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>

Route: <sip:172.23.22.49:5080;transport=udp;lr>

Session-Expires: 1800;refresher=uac

Allow: INVITE,BYE,ACK,CANCEL,OPTIONS

Content-Type: application/sdp

Content-Length: 315

v=0

o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104

s=SIP Call

c=IN IP4 172.23.12.104

t=0 0

m=audio 30248 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

Can somebody please if it is a FS configuration problem or a software
problem or is it a problem on the sipx side?

Thanks in advance,

Nikolay.

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