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Unread 06-28-2008, 04:18 AM   #1
wishfull
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Default No sound on PSTN access numbers

I have SIP phone and for few weeks now, when people call using PSTN or IM access numbers we can not hear each other. VOIP to VOIP call work without any issues. Can some please help.
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Unread 06-28-2008, 04:46 PM   #2
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Originally Posted by wishfull View Post
I have SIP phone and for few weeks now, when people call using PSTN or IM access numbers we can not hear each other. VOIP to VOIP call work without any issues. Can some please help.
-Which PSTN Access number is the one i question?
-Does your SipPhone have an option to set up STUN?
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Unread 05-31-2009, 09:05 PM   #3
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Quote:
Originally Posted by emoci View Post
-Which PSTN Access number is the one i question?
-Does your SipPhone have an option to set up STUN?
I hope it's okay to bring up this older thread, as I'm hitting the issue as well. I cannot hear the PSTN party, but he can hear me. This is via the Toronto, Canada access number, the only city of my interest at this time. STUN is not an option because I am behind a NAT, which is running the DD-WRT router firmware (IP Tables Symmetric NAT), plus the Nokia WiFi phone that I'd like to use at times does not support STUN anyway.

Based on some sniffing, I see the SIP signaling is coming in from IP 204.11.194.10. I am asked via an SDP message to send my RTP stream to that same address, which I do, though I do not receive any RTP traffic from it. A few seconds later, I start receiving RTP data from IP 64.34.164.254, then I am asked (a 2nd SDP) to redirect my RTP stream to this new IP. I honor the redirect.

On the LAN side of things, everything looks solid, but I noticed that the source port number of my 2nd outbound RTP socket was not properly preserved by my router. Somehow it was not preserved even though it is clearly a new socket, yet the disqualification must relate to the fact that the same source port was used with the 1st RTP socket (and which is still active in the router). The lack of port preservation means that the Symmetric RTP relationship has been broken, and I am left stuck with one-sided audio.

Obviously the lack of any easy holistic solution makes this frustrating. But as I think about this, I can't be alone, as there are going to be NAT devices out there that don't always do the right thing (e.g., preserve source ports). Therefore, in the interest in maximizing success for all parties, I wonder if the SIP Broker techs could see if it might be possible to configure the server such that the very first SDP message contains the correct media IP address (64.34.164.254, at least my case), so that there would not need to be any kind of call transfer process once the media has been negotiated.

Thank you for checking!
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Unread 06-01-2009, 05:39 PM   #4
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A few things to try:

Outgoing Calls: Can you reach *600 (Echo Test)

Incoming Calls:

-Try calling a SIPBroker number (SIPBroker - PSTN Numbers), then *010123456 , 123456 replaced with your VoXalot number. Do you receive this call, and is audio ok?

-Try entering your voxalot URI 123456@voxalot.com in the link below. Instead of you calling Echo, echo will call you so you can see how it behaves when it is an incoming call: SIPBroker - EziDial

Possible Solutions:

-If your router has UPnP I would suggest activating that rather than DMZ

-Preferably make sure you have STUN set up at least on the Device that supports it (stun.xten.com has been working for me). STUN is definitely an option, and especially useful behind NAT (although I realize that it is not an option for certain devices)(

-This is optional, but it may help to open these port ranges and forward them to your ATA's IP:

5050-5064
5000-5005
16300-16500 (this maybe slightly different for your ATA, there should be an RTP port range setting on your ATA, if it is different, note the range and open that range instead in your router...)
For some background on this see: http://forum.voxalot.com/12685-post20.html

-Make sure there are no Internet Connection problems. Run a SpeedTest, or maybe VoIP test at TestYourVoIP.com

-See if this sparks any ideas http://forum.voxalot.com/voxalot-sup...t-routers.html

-Lastly what are the devices involved on the two ends...I remember reading previous issues with the Toronto Number and SPA3102 interaction ...

The issue is that the Access numbers are donated from various providers. Although they all terminate to SipBroker servers ... in some cases the actual underlying provider plays a part in the connection (and this falls beyond what SipBroker can control and adjust)...

One last thing to attempt is to use a Toronto Access number from one of these alternatives and see if the problem persists:
http://forum.voxalot.com/sip-broker-...r-gateway.html

Finally....if you happen to have a regular DID anywhere else in Canada you may just want to grab a Toronto Number to forward to your current DID from FreePhoneLine.ca (this will not be shared but your own DID, and free as long as you just want to forward it...)
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Unread 06-06-2009, 11:24 PM   #5
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Quote:
Originally Posted by emoci View Post
A few things to try: ...
Thank you, Emoci! I understand that it may not be possible to make adjustments on the PSTN numbers, as different firms may operate different ones. And I believe the SIP call-transfer behavior from the service is valid, but was too bad that it exposed this NAT issue locally. Based on some further checking, it appears that DD-WRT (with its Netfilter NAT logic) does not yet support the RFC 4787 best practice of "Endpoint-Independent Mapping" (http://www.rfc-archive.org/getrfc.php?rfc=4787).

I did not try the other PSTN numbers as their dialing arrangements weren't as direct, but for completeness did try the echo test (sip:*010*600@sipbroker.com) and EziDial. The echo call set up and my RTP stream went out, but no incoming on the LAN side; I didn't probe the WAN side, but suspect the far end did not set up as Symmetric RTP. The EziDial did not work (no incoming call at all), but I tried with a generalized SIP address as I do not have a Voxalot number. However, the sip:*850301@sipbroker.com and sip:1234@loligo.com echo tests work fine here.

Fortunately I'm now in business with the PSTN number. This Nokia E63 phone always sets up its RTP stream with a source port number of 49152, and since that matches the target port of SIP Broker's stream to me, I set up a UDP port forwarding range rule (49152-49155) direct to the phone, and am now up. When at a public hotspot, I may need to switch to the Fring application (RTP media proxying), and that's a fine compromise.

Thanks again and I hope this helps others as well.

ps: I just tried the *600 echo test again, this time on the Nokia (was on the computer) with its port forwarding rule. It ran fine.
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Unread 06-13-2009, 11:45 PM   #6
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Quote:
Originally Posted by emoci View Post
A few things to try: ...
Hmm...my earlier reply never got posted, so hoping this makes it. Thanks Emoci for the tests and additional methods.

On echo tests, sip:1234@loligo.com and sip:*850301@sipbroker.com work in all contexts. The one at sip:*010*600@sipbroker.com though only works when the appropriate SIP headers contain a public IP address. So to ensure maximum compatibility, one's client should be able to obtain his public IP (probably via STUN) and announce with it, and/or one should manage outgoing calls through a registered account (such as MySIPSwitch.com). The call-me echo test did not work, probably because I am not on a Voxalot account.

I have the PSTN gateway playing now. I finally determined that the call transfer itself was not the problem, but was because the inbound RTP stream beat my new outgoing RTP stream, thus grabbing the router's IP tables connection that I needed. The solution: turn on the SPI firewall in DD-WRT, so that any RTP stream that beats mine is dropped before it enters. (The firewall is on by default, but I thought I knew better and had it off.) Forwarding RTP ports is an alternative if one really has to have the firewall off.

The Toronto PSTN number works very well, but is time limited to just over 3 minutes. I also tested TPad (the number works, but account is no good) and Bezecom (works, but not yet sure of maximum call duration).

Thanks again, and hope this helps others!
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Unread 06-14-2009, 12:09 AM   #7
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Originally Posted by Hal View Post

The Toronto PSTN number works very well, but is time limited to just over 3 minutes.
Hmm,

That's weird...I'll actually test this...

It is no secret that I have my acct. setup for CallThru via PBXes, and I use the Toronto Access Number for CallThru once in a while without issues in terms of time limits...but I'll test anyways...
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