[Asterisk-Users] SIP Testing

[Eric Wieling]

I would like to get my Asterisk server working with Packet8.  I already
have my DTA310 talking to Asterisk just fine.

What did you do to get Asterisk working with Packet8?  Are you behind a
NAT firewall?  I am and was not able to get it working with an Asterisk
built from CVS a couple of weeks ago.

I'm behind a NAT firewall.

Thanks in advance for any help!

--Eric

On Mon, 2003-04-07 at 11:03, Stephen Davies wrote:
> On Sun, 6 Apr 2003, Mark Spencer wrote:
> 
> > In general, I find that SIP is extremely fragile, and every time I try to
> > fix one bug, I end up creating another somewhere.  What I need are
> > strategies for verifying that the SIP implementation is correct, either
> > via some sort of SIP test suite or even just a collection of users who
> > will sign off on things.
> > 
> > Anyway I'm soliciting for ideas from the list.  I'd be happy to get some
> > feedback.
> 
> Well - I did some testing with the the current CVS.
> 
> I tested with:
> 
> 1) As local client: a Cisco ATA186, both ports configured as local
> "friends" of * (extn 6001 and 6002)
> 
> 2) As "remote" SIP call targets or sources:
> 
> a) On Free World Dialup:
> 
>  - An SJPhone client, using FWD's proxy service for getting through NAT,
>    FWD number 21622
>  - The Libertel eDial conference server on FWD 14551
> 
> b) On Packet8:
> 
>  - Packet8's DTA310 SIP adapter (like ATA186), using Packet8's broadband
>    phone service (www.packet8.net) [I've enabled g711 on my DTA310]
> 
> My Asterisk registers with FWD with my FWD number 21542.
> 
> On my setup reinvites are turned off - my ATA186 at home is on an unrouted
> address so "native bridging" between them and outside SIP services won't
> work.
> 
> I made the following tests.  In every case I check that the call cleared
> correctly from either end.
> 
> 
> Test 1: "intercom calls" from port 1 of ATA to port 2, via Asterisk
> 
>   A simple setup - no proxies involved.
> 
> 					>> Test PASSED
> 
> Test 2: outgoing call from * to FWD, calling the SJPhone mentioned above
> 
>   For calls via FWD to work, Record-Route handling needs to be done
>   right.  My SJPhone client is configured to work through FWD's
>   Peerpoint NAT proxy  by Jasomi Networks - so SIP traffic passes
>   through 2 proxies, RTP streams also pass through the Jasomi Peerpoint.
> 
> 					>> Test PASSED
> 
> Test 3: outgoing call from * to FWD, calling the Libertel eDial conference
> system on FWD 14551
> 
>   The Libertel conference system is reached through FWD, so again
>   Record-Route handling must work.  Doesn't use the Peerpoint, though.
> 
>   In this case I couldn't test clearing the call from the eDial end - I
>   don't have control of that end, and their IVR wouldn't hang up on me.
> 
> 					>> Test PASSED
> 
> Test 4: outgoing call from * to my Packet8 account,
> SIP/1847xxxyyyy at packet8.net
> 
>   Packet8 will see this as a call from "outside" their network.
> 
>   In this test call * did quite a few retransmits before the Packet8
>   service started to respond.  So it exercised the retransmit code.
> 
> 					>> Test PASSED
> 
> Test 5: incoming call from SJPhone client on FWD to my
> 21542 at fwd.pulver.net
> 
>   Inbound from SJPhone via the Jasomi Peerpoint and the FWD proxy.
> 
>   BUG: BYE originated from * end was not seen at SJPhone - lost
>   in transit.  SJPhone (obviously) didn't OK it.  But * did not
>   retransmit.  Call did not clear at  the SJPhone end.  The
>   bug is the lack of retransmits - on subsequent
>   tests where the BYE wasn't lost the call cleared fine.
> 
>   BUG? Asterisk does not return the Record-Route header in the
>   "180 Ringing" response.  FIXME: Check against RFC!!.  Didn't
>   affect the call as far as I can tell.
> 
> 					>> Test FAILED
> 					>> (though call "worked")
> 
> 
> Test 6: incoming call from the PSTN to 21542 at fwd.pulver.com, via eDial
> test inbound gateway
> 
>   From *'s point of view, like Test 5 except that the Peerpoint proxy
>   isn't used.
> 
>   Same bug with Record-Route not copied back in 180 response seenm
>   but apart from that:
> 
> 					>> Test PASSED
> 
> 
> 
> So that set of tests were mostly rather successful.  Two bugs found:
> 
> BUG#1: BYE not retransmitted - for Mark to fix I'd say.
> 
> BUG#2? Record-Route not copied back to 180 response - for me to
> investigate, maybe fix
> 
> 
> Regards,
> Steve
> 
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