[Freeswitch-users] SIP Trace Question / NAT one way audio problem.

Sean P Devoy sdevoy at bizfocused.com
Fri Dec 6 07:27:42 MSK 2013 

Hi Avi,

 

Thanks for the reply, nice to "hear" from you again.  You too Ken.  FYI: I
do have NDLB force rport set to TRUE, I will change to safe.

 

My Cisco 504G has been working like a champ for like a year or so, and I
just noticed in the sip trace it appears to REGISTER every 23 seconds.
Maybe I am reading this wrong.  I don't see any NAK w/ nonce and
re-REGISTER.  Also, the CSeq DOES increment, so it appears to have gotten
the response.  It seems to me the CSeq stays the same if the sender does not
get a response at all.

 

Here is my SIP trace extract for a WORKING CISCO 504G:

 

recv 757 bytes from udp/[7x.1xx.1xx.9x]:5063 at 17:53:12.164203:

------------------------------------------------------------------------

REGISTER sip:xyz.mydomain.com SIP/2.0

Via: SIP/2.0/UDP 7x.1xx.1xx.9x:5063;branch=z9hG4bK-238fcc8b;rport

From: "BFIS" <sip:220 at xyz.mydomain.com>;tag=163202d2fcedcf5eo0

To: "BFIS" <sip:220 at xyz.mydomain.com>

Call-ID: 92b82661-3cbfddb2 at 10.10.40.50

CSeq: 627972 REGISTER

Max-Forwards: 70

Authorization: Digest
username="220",realm="xyz.mydomain.com",nonce="7ccf6076-769a-4857-9f53-30808
b2a4e61",uri="sip:xyz.mydomain.com",algorithm=MD5,response="b3ec391477f137bf
af2393c97b4e0f21",qop=auth,nc=0000010e,cnonce="49035914"

Contact: "BFIS" <sip:220 at 7x.1xx.1xx.9x:5063>;expires=3600

User-Agent: Cisco/SPA504G-7.5.4

Content-Length: 0

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

------------------------------------------------------------------------

 

send 615 bytes to udp/[7x.1xx.1xx.9x]:5063 at 17:53:12.168173:

------------------------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/UDP 7x.1xx.1xx.9x:5063;branch=z9hG4bK-238fcc8b;rport=5063

From: "BFIS" <sip:220 at xyz.mydomain.com>;tag=163202d2fcedcf5eo0

To: "BFIS" <sip:220 at xyz.mydomain.com>;tag=pS3teKg2aNtgj

Call-ID: 92b82661-3cbfddb2 at 10.10.40.50

CSeq: 627972 REGISTER

Contact: <sip:220 at 7x.1xx.1xx.9x:5063>;expires=30

Date: Wed, 04 Dec 2013 17:53:12 GMT

User-Agent: FreeSWITCH-mod_sofia/1.2.9+git~20130518T213043Z~4fc1369a1b

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

Supported: precondition, path, replaces

Content-Length: 0

------------------------------------------------------------------------

 

send 769 bytes to udp/[7x.1xx.1xx.9x]:5063 at 17:53:13.987120:

------------------------------------------------------------------------

OPTIONS sip:220 at 7x.1xx.1xx.9x:5063 SIP/2.0

Via: SIP/2.0/UDP 66.241.102.41;rport;branch=z9hG4bKZvvNXpHcSeKQS

Max-Forwards: 70

From: <sip:mod_sofia at 66.241.102.41:5060>;tag=1vSB2v98aQFyH

To: <sip:220 at xyz.mydomain.com>

Call-ID: 5f13a1c3-814d-4f48-837a-b906e5288b00_92b82661-3cbfddb2 at 10.10.40.50

CSeq: 61471088 OPTIONS

Contact: <sip:mod_sofia at 66.241.102.41:5060>

User-Agent: FreeSWITCH-mod_sofia/1.2.9+git~20130518T213043Z~4fc1369a1b

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

Supported: precondition, path, replaces

Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer

Content-Length: 0

------------------------------------------------------------------------

 

recv 451 bytes from udp/[7x.1xx.1xx.9x]:5063 at 17:53:14.054668:

------------------------------------------------------------------------

SIP/2.0 200 OK

To: <sip:220 at xyz.mydomain.com>;tag=dcdb8ee2502b328ei0

From: <sip:mod_sofia at 66.241.102.41:5060>;tag=1vSB2v98aQFyH

Call-ID: 5f13a1c3-814d-4f48-837a-b906e5288b00_92b82661-3cbfddb2 at 10.10.40.50

CSeq: 61471088 OPTIONS

Via: SIP/2.0/UDP 66.241.102.41;branch=z9hG4bKZvvNXpHcSeKQS;rport=5060

Server: Cisco/SPA504G-7.5.4

Content-Length: 0

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

------------------------------------------------------------------------

 

recv 412 bytes from udp/[7x.1xx.1xx.9x]:5063 at 17:53:14.733495:

------------------------------------------------------------------------

NOTIFY sip:xyz.mydomain.com SIP/2.0

Via: SIP/2.0/UDP 7x.1xx.1xx.9x:5063;branch=z9hG4bK-b8f05783;rport

From: "BFIS" <sip:220 at xyz.mydomain.com>;tag=163202d2fcedcf5eo0

To: <sip:xyz.mydomain.com>

Call-ID: f5c9e863-5d603f02 at 10.10.40.50

Max-Forwards: 70

Contact: "BFIS" <sip:220 at 7x.1xx.1xx.9x:5063>

Event: keep-alive

User-Agent: Cisco/SPA504G-7.5.4

Content-Length: 0

------------------------------------------------------------------------

 

send 741 bytes to udp/[7x.1xx.1xx.9x]:5063 at 17:53:14.733877:

------------------------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/UDP 7x.1xx.1xx.9x:5063;branch=z9hG4bK-b8f05783;rport=5063

From: "BFIS" <sip:220 at xyz.mydomain.com>;tag=163202d2fcedcf5eo0

To: <sip:xyz.mydomain.com>;tag=rcZ5aBDt83Fcc

Call-ID: f5c9e863-5d603f02 at 10.10.40.50

Contact: <sip:gw+didlogic at 66.241.102.41:5060;transport=udp;gw=didlogic>

User-Agent: FreeSWITCH-mod_sofia/1.2.9+git~20130518T213043Z~4fc1369a1b

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

Supported: precondition, path, replaces

Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer

Content-Length: 0

------------------------------------------------------------------------

 

recv 412 bytes from udp/[7x.1xx.1xx.9x]:5063 at 17:53:29.805194:

------------------------------------------------------------------------

NOTIFY sip:xyz.mydomain.com SIP/2.0

Via: SIP/2.0/UDP 7x.1xx.1xx.9x:5063;branch=z9hG4bK-ef6775c6;rport

From: "BFIS" <sip:220 at xyz.mydomain.com>;tag=163202d2fcedcf5eo0

To: <sip:xyz.mydomain.com>

Call-ID: f5c9e863-5d603f02 at 10.10.40.50

Max-Forwards: 70

Contact: "BFIS" <sip:220 at 7x.1xx.1xx.9x:5063>

Event: keep-alive

User-Agent: Cisco/SPA504G-7.5.4

Content-Length: 0

------------------------------------------------------------------------

 

send 741 bytes to udp/[7x.1xx.1xx.9x]:5063 at 17:53:29.805669:

------------------------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/UDP 7x.1xx.1xx.9x:5063;branch=z9hG4bK-ef6775c6;rport=5063

From: "BFIS" <sip:220 at xyz.mydomain.com>;tag=163202d2fcedcf5eo0

To: <sip:xyz.mydomain.com>;tag=Dajeg3e4BZ3mF

Call-ID: f5c9e863-5d603f02 at 10.10.40.50

Contact: <sip:gw+didlogic at 66.241.102.41:5060;transport=udp;gw=didlogic>

User-Agent: FreeSWITCH-mod_sofia/1.2.9+git~20130518T213043Z~4fc1369a1b

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

Supported: precondition, path, replaces

Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer

Content-Length: 0

------------------------------------------------------------------------

 

recv 757 bytes from udp/[7x.1xx.1xx.9x]:5063 at 17:53:35.236173:

------------------------------------------------------------------------

REGISTER sip:xyz.mydomain.com SIP/2.0

Via: SIP/2.0/UDP 7x.1xx.1xx.9x:5063;branch=z9hG4bK-4d49f459;rport

From: "BFIS" <sip:220 at xyz.mydomain.com>;tag=163202d2fcedcf5eo0

To: "BFIS" <sip:220 at xyz.mydomain.com>

Call-ID: 92b82661-3cbfddb2 at 10.10.40.50

CSeq: 627973 REGISTER

Max-Forwards: 70

Authorization: Digest
username="220",realm="xyz.mydomain.com",nonce="7ccf6076-769a-4857-9f53-30808
b2a4e61",uri="sip:xyz.mydomain.com",algorithm=MD5,response="f0da4a7351b8d978
20cfa98a1669a695",qop=auth,nc=0000010f,cnonce="49035914"

Contact: "BFIS" <sip:220 at 7x.1xx.1xx.9x:5063>;expires=3600

User-Agent: Cisco/SPA504G-7.5.4

Content-Length: 0

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

------------------------------------------------------------------------

 

And so it continues.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Thursday, December 05, 2013 10:55 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP Trace Question / NAT one way audio
problem.

 

I set the linksys spas to have a ping of 15 seconds, but not to re register
every 15 seconds!

When I see constant register attempts, it always means that the 401 was not
received so the register never completes. Are these phones actually
successfully registered?

-Avi

On Dec 6, 2013 4:47 AM, "Ken Rice" <krice at freeswitch.org
<mailto:krice at freeswitch.org> > wrote:

Re-registering every 23 seconds isnt that big of a deal if the phones are
behind nat, sometimes this is needed due to the way NAT happens on UDP with
various routers. There is no way to track the start/end of the session w/out
using an ALG. I have seen some NAT routers require a re-reg interval around
15 seconds which is a bit excessing

On the issue w/ the non-working ones is you'll notice the rport on the one
that's working...

On the ones that arent working they arent specifying rport... If those are
the polycoms look for NDLB force rport setting and set it to safe... That's
a special mode for the polycoms...

Check the wiki for more NAT handling examples
K


On 12/5/13 8:36 PM, "Sean P Devoy" <sdevoy at bizfocused.com
<http://sdevoy@bizfocused.com> > wrote:


Hi all,
 
I seem to have fallen off the list!  No mail since 7/26/2013.  Have I made
someone mad??  :)

I have also been blissfully running with ISSUES what so ever since that
time.
 
I have a NATed client with PERIODIC one way audio.  The client end has a
Cisco 220W router with SIP ALG disabled.  The phones are CISCO 504Gs and
Polycom 330s(?), basciallt the same as everywhere else we have phones.
 
I did a sofia global sip trace on for a while and captured the output.
After writing an app to sort those sip packets into logical stream files, I
dug in to the SIP conversations.  This has left me with som questions for
the people who know these things:
 
1.   I found all my phones are REGISTERing every 23 seconds!   That seems
absurd TO ME, so where have I screwed that up/where do I set it and what is
a reasonable number?

2.   The only difference I see in the SIP packets from my working phone and
these non-working ones is of course NAT related.  The working phone's
REGISTER looks like this:

recv 757 bytes from udp/[WW.XX.YY.93]:5063 at 17:53:12.164203:
------------------------------------------------------------------------ 
REGISTER sip: <MYDOMAIN.COM <http://MYDOMAIN.COM> > SIP/2.0
Via: SIP/2.0/UDP WW.XX.YY.93:5063;branch=z9hG4bK-238fcc8b;rport

But the failing phones look like this:



recv 801 bytes from udp/[AA.BB.CC.38]:5104 at 17:53:40.693691:

------------------------------------------------------------------------

REGISTER sip: <MYDOMAIN.COM <http://MYDOMAIN.COM> > SIP/2.0

Via: SIP/2.0/UDP 10.2.2.239:5104;branch=z9hG4bKb374d6507894B431



The response is still sent to [AA.BB.CC.38]:5104


I guess first, IS THAT A PROBLEM?
 
So, can anyone share any insight into the CISCO 220W config that I might be
missing?  
 
Why would it only be a periodic problem?  Could this be a problem where one
end is using a wider range of rports than the other supports?
 
Thanks,
Sean
 




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