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08-21-2008, 10:19 PM | #1 |
Junior Member
Join Date: Aug 2008
Posts: 16
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Problems with Symmetric NAT
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. |
08-22-2008, 07:26 PM | #2 |
Member
Join Date: May 2008
Posts: 80
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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... Last edited by TheFug; 08-22-2008 at 07:29 PM. Reason: btw... |
08-23-2008, 01:50 AM | #3 | |
Junior Member
Join Date: Aug 2008
Posts: 16
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@TheFug:
FWD? No, Jeff Pulver says this: Quote:
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. |
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08-23-2008, 05:22 PM | #4 |
Senior Member
Join Date: Jul 2007
Location: Oregon, USA
Posts: 365
Thanks: 17 Thanked 77 Times in 64 Posts |
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. Last edited by boatman; 08-24-2008 at 07:29 AM. |
08-23-2008, 07:48 PM | #5 |
Join Date: Jul 2007
Location: Toronto, Canada
Posts: 1,422
Thanks: 123 Thanked 369 Times in 282 Posts |
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...) |
08-24-2008, 07:10 AM | #6 | ||||||||
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Join Date: Aug 2008
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Thanks boatman, emoci.
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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. Quote:
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. Last edited by jdstroy; 09-05-2008 at 01:21 AM. Reason: More test results: Phoner for Windows |
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08-30-2008, 04:49 AM | #7 |
Junior Member
Join Date: Aug 2008
Posts: 16
Thanks: 6 Thanked 0 Times in 0 Posts |
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? Last edited by jdstroy; 09-05-2008 at 01:25 AM. Reason: Verified, TCP used in Phoner |
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