Problems with Symmetric NAT

[jdstroy]

Dear Voxalot community,

I've finally decided to join Voxalot after hearing about the no-longer-free Free World Dialup. FWD had a complete and working setup for symmetric NAT, and I used that for a long time. But now that August is coming close to an end, it appears that I have to move on to another provider with symmetric NAT support (e.g. with RTP media proxies). So here I am.

Things are nice without a symmetric NAT. On a port-restricted cone NAT, things are fine. However, I can't always have a decent connection, and now that I'm back in college, the network connection is horrible (because the people who run the network are arse-holes). I'm stuck behind a symmetric NAT, and there's no way (for me) to do port forwarding.

I'm currently experiencing the ever common "Waiting for acknowledgment / Ack timeout" and a few half-legged (e.g. outbound only) calls, and occasionally, an altered/mismatched codec call (usually gsm inbound, and PCMU outbound); all of these, of course, are not functional. The inbound audio is often missing, but sometimes it works correctly when it decides to route the PCMU inbound stream correctly.

So, my new friends and neighbors at Voxalot, I'd like to know if there's anything I can do to actually get decent call connections. The UA that I'm using is SJPhone 1.65.377a. These are the configuration settings that I'm using:

Outgoing calls:
Session initialization timeout: 0 sec

NAT mapping refresh:
Method: OPTIONS request
Timeout: 10 sec

Profile: Voxalot
Account: ISR
Password: ISR
(ISR = Inquired, Saved, Required)

Domain/Realm: voxalot.com
Use Outbound Proxy: Yes
Proxy URI: sip:us.voxalot.com
Proxy usage model: Strict outbound proxy

Use separate Outbound Proxy for NAT: Yes
Proxy URI: sip:us.voxalot.com
Proxy usage model: Strict outbound proxy

.... [snip -- most of these are irrelevant] ...

Use rport extensions: yes
Voicemail number: 500

.... [snip -- most of these are irrelevant] ...

Use discovered addresses in STUN: No

End.

Any suggestions that I could try to get SJPhone with Symmetric NAT working? I'd happily provide a log if that would assist in diagnosing the problem and solving the issue.

Thanks,

J

PS: If anyone knows of another SIP provider with symmetric NAT support, I'd definitely like to know, too. PM me as needed. I'm desperate to try anything at this point.

[TheFug]

Don't know exactly, but look for for Brujula or IXCALL

btw. you can still use their free service, after August....peering wil also be available...

[jdstroy]

@TheFug:

FWD? No, Jeff Pulver says this:

If you wish to retain SIP registration and support FWD, please click through the paid membership banner at the top of the home page which links to the Acteva registration service.

Peering, of course, is available, but I could care less about peering. What I really want is a working RTP proxy.

But, thanks for the suggestion of IXCALL! I will give it a try.

Does anyone else have any suggestions? It would be nice if I could avoid registering for another service and just get Voxalot SIP on symmetric NAT working.

[boatman]

If you are behind a Symmetric NAT you may have to send all packets (including RTP packets) through a proxy.

Ordinarily Voxalot will try to have the RTP packets going directly between endpoints, however if the party receiving the call has incorrect NAT settings then Voxalot will proxy the RTP packets. I have not tested this, but the setting at Voxalot "Symmetric NAT Handling" = yes, and "Optimize Audio Path" = no, should cause Voxalot to proxy the RTP packets.

Assuming you can somehow get Voxalot to proxy the RTP packets, the connection may sound better if you use one of the low bit rate codecs such as G729.

[emoci]

A few things:

-Take a look at this, and see if it helps Setting up your SJ Phone softphone to use Voxalot - Voxalot FAQ

-Considering your situation, it makes sense to me to have:
Symmetric NAT handling (under Member Details on VoXalot) set to Yes + STUN server if possible on SJPhone

-Also take a look at this (just steps 1 & 2 there): HowTo: 6 Steps To Optimize Your Audio - Voxalot FAQ


Are you really attached to SJPhone, if not I would try XLite and/or Zoiper (I've come to like Zoiper when out of the house, it is very straigtforward to setup, and I've had good results so far...)

[jdstroy]

Thanks boatman, emoci.

Originally Posted by boatman

If you are behind a Symmetric NAT you may have to send all packets (including RTP packets) through a proxy.

Yep, knew this; still having trouble with this.

Originally Posted by boatman

Ordinarily Voxalot will try to have the RTP packets going directly between endpoints, however if the party receiving the call has incorrect NAT settings then Voxalot will proxy the RTP packets, so you should slightly mis-configure your NAT settings.

Isn't this what the Symmetric NAT option is for? My network is, after all, really messed up already. I hate using the University network.

Originally Posted by boatman

There may also be other instances when Voxlaot proxies RTP packets. I have not found any written information about this. There is a setting in my Voxalot web interface regarding NAT, but I don't know what that does.

I believe it's for what I want it to do -- proxy RTP traffic. (It says, "Set to 'Yes' to enable Voxalot NAT handling logic.) It would be helpful if I could figure out why the traffic isn't handle properly.

Originally Posted by boatman

Assuming you can somehow get Voxalot to proxy the RTP packets, the connection may sound better if you use one of the low bit rate codecs such as G729.

I will try G729, as well as iLBC; however, I don't think that this is the issue at hand, though. Thanks for the tip. (As for now, I just want the connections to go through correctly. Sound quality will be important after I actually get the connections working properly.)

Originally Posted by emoci

A few things:

-Take a look at this, and see if it helps Setting up your SJ Phone softphone to use Voxalot - Voxalot FAQ

Did that already, yep.

Originally Posted by emoci

-Considering your situation, it makes sense to me to have:
Symmetric NAT handling (under Member Details on VoXalot) set to Yes + STUN server if possible on SJPhone

I did the prior, but I don't think STUN would help on SJPhone. I recall reading that SJPhone's STUN feature is broken. Not to mention, I'm definitely behind a symmetric NAT that definitely doesn't play well with STUN. (Is there a way to configure TURN on SJPhone? I did find a TURN server.)

Originally Posted by emoci

-Also take a look at this (just steps 1 & 2 there): HowTo: 6 Steps To Optimize Your Audio - Voxalot FAQ

I assume you mean that I should leave Symmetric NAT handling to Yes. I've been using the service for about half a week now, and I know that inbound definitely doesn't work properly. Both inbound and outbound calls show "Awaiting Acknowledgment", which progresses to "(out: PCMU)" in later calls, and then finally to "(PCMU)."

To clarify:

All calls made or received after a certain period of idle time (timeout) will show "Awaiting Acknowledgment" in SJPhone's status.

All calls made or received subsequently may either show "Awaiting Acknowledgment" or "(out: PCMU)," which means that one leg of the call is missing (lost somewhere between me and Voxalot).

Calls established successfully indicate "(PCMU)" in the status, which means that both legs worked properly. These occur most often after several consecutive inbound or outbound calls that fail, and calls immediately after often succeed.

The echo test does seem to work better than other calls. I don't know if this is just my perception, or if it does indeed work better.

Originally Posted by emoci

Are you really attached to SJPhone, if not I would try XLite and/or Zoiper (I've come to like Zoiper when out of the house, it is very straigtforward to setup, and I've had good results so far...)

I really do like SJPhone because it has all the features that XLite doesn't. It feels (and works) more like a power VoIP utility than XLite (e.g. direct SIP URIs, any number of profiles, any number of lines, conference calls without a limit on attendees...). Plus, it works better on my network. I've used XLite before, and it didn't work very well at my location (works much better at home). However, XLite requires less configuration, which can be a good thing. I'll give Zoiper a try when I get back on site.

Thanks again, boatman and emoci.

(Does anyone know why forum posts take so long to show up, and why they sometimes don't even show up at all?)

Okay, I've tested with X-lite, and the results are more disappointing than SJPhone.

Calling 18004664411 (Goog 411), Xlite experiences the "Awaiting Acknowledgment" problem. I didn't mention it before, but part of the incoming audio is received while experiencing this issue. For example, calling Goog 411, the problem sometimes results in this:

[Ring] "Ca-" [silence].

It should be "Calls recorded. Goog 411!" In SJPhone, you may sometimes hear "Ca-" followed by silence, timeout, and session teardown, while in XLite, you may hear "Ca-" followed by silence, and finally "Say just a city and state, like San Francisco, California."

--For the record, 600, the echo test, works properly in Xlite and SJPhone.--

^ No longer true. 600 doesn't always work, but it works better than other calls.

Additionally, I have a DID that I use for incoming calls from IPKall. It, too, experiences a similar problem.


I've also tried Phoner for Windows. It experiences the same issue.

[jdstroy]

Is there a way to look at server-side logs on RTP connections that I make, or that are made for me? (Or, better yet, is there a way to force proxied RTP connections?)

It seems the largest problem for me is the "Awaiting Acknowledgement" issue. I've switched to SIP/TCP instead of SIP/UDP, and it still does the same thing, so I'm guessing it's more of a problem with the media streams than the signalling streams.


I've used Phoner with both SIP/TCP and SIP/UDP registrations. I'm almost certain it's a RTP proxy issue; when I use Wireshark, I can see normal connections, which have both incoming and outgoing RTP audio data, and I can see problematic connections, which only have outgoing RTP media and no incoming media.

Is there something I can do to force the use of the RTP proxy?