Whatever fixed this is still in the final release of 4.4, so I am very
happy. I can upgrade my sites from 4.2.1 to 4.4 and start using G729
again instead of G711.
Thanks again Ezuce team!
On 1/28/2011 5:11 PM, Matthew Kitchin (public/usenet) wrote:
Thanks to Matt White, I know a lot more about this issue.
It is/was not related to the port translation issue. The phones were
doing that because of this bug:
http://track.sipfoundry.org/browse/XTRN-942
They do not do it when they are running 3.2.4.
We spent a lot of time troubleshooting why my issue was happening.
Something was causing Verizon to drop the call.
We suspected it may have been because of this:
http://www.mail-archive.com/[email protected]/msg15075.html
partially because this jira item was specifically opened for a similar
issue with Verizon:
http://track.sipfoundry.org/browse/XX-7517
I loaded up a 4.4 machine to make sure I could test this option:
<strict-protocol-enforcement>false</strict-protocol-enforcement>
mentioned here:
http://www.mail-archive.com/[email protected]/msg15075.html
but I didn't need to. Everything works in 4.4. I told sipxbridge to
allow all codecs, and it works perfectly.
The call starts out at G729, drops to G711 for hold, changes back to
G729 when the transfer is done.
It works perfect. Oh well. Something must have changed...
On 1/27/2011 1:04 PM, Matthew Kitchin (public/usenet) wrote:
On 1/27/2011 12:57 PM, Michael Scheidell wrote:
On 1/27/11 1:39 PM, Matthew Kitchin (public/usenet) wrote:
I can shed some light on that I think. Verizon required they send
us calls on port 5060. We do nat the port to 5080. We hadn't seen
any issue there, but I guess we have now for some reason.
Verizon --5060-- our router -- 5080 --> sipx
when I had this problem with level3, and TRIED everyone's
suggestions of pat, port forwarding, etc, I found that at LAYER4 IT
WORKED, but phucked up at LAYER7.
this won't work.. did it work? no, it just sorta worked. it won't
work. as in 'undefined', as in maybe, under certain circumstances,
sometimes it works.
you need an application layer proxy. you need a seperate copy of FS
to do this. you need an SBC to do this. you need a simple perl
script to do this, and/or, I think the juniper firewalls can modify
sip invites on the fly.
as in, what needs to be done?
verizon port 5060 (in the invite, verizon sends the '5060' somewhere)
you neet to xlate to port 5080. not only layer4, but layer7
(netsed?). sipx sees it as coming TO port 5080, and responds back,
saying 'thanks, I got this on port 5080, continue on port 5080'
the OUTBOUNDS needs to xlate src port 5080 on sipx to 5060, AND THEN
TRANSLATE THE ASCII TEXT 5080 BACK TO 5060.
so, in your case, what you are trying to do, won't work.
no, don't tell me it sorta works, you just proved it won't work.
Tony and I tried a while back, and I tried with a copy of FS on
pfsense, then finally convinced level3 that if they want my $17 a
month, they needed to send to me on port 5080.
Tony can tell you about how hard level3 will work for your business.
you might call them. tell them tony and I send you.
(or help us come up with the bounty to pay for a good pfsense/fs
config that will do the proxying.. I still think a couple of copies
of netsed might do it, but I don't need to try today)
how to try cheaply?
get voip.ms to work with static ip authentication. if you get that
to work, then you have it.
I'm not sure if that is causing my problem or not. We do not have
remote workers, and I do understand where it would cause a problem
there. I cannot use voip.ms here. This is a 110 site MPLS network
that runs on Verizon infrastructure. For a variety of reasons, I need
to use Verizon Voip services. I don't believe I have proved that the
port translation is or isn't causing it.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
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