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--- RE: pap2t - help - random hangups :( (http://www1.pbxes.com/forum/threadid.php?threadid=1255557848)

help - random hangups :(

Hi,

I am attempting to track down what is happening with my calls.

My setup:
pap2t->router (tomato 1.25)->att dsl (bridge mode)->internet

At random times, from 30s to 30min to never, I get random OUTBOUND audio drops. The connection is still active, but outbound audio (from the pap2t to the PSTN) has stopped for some reason.

The caller ALWAYS complains about a loud DTMF/FAX tone when the audio drops. I can hear them, they can't hear me. This happens on both sent and received calls.

Do you have any idea what would cause a loud DTMF tone right before the outbound audio is dropped?

I currently have the pap2t in the DMZ so it's not port forwarding issue. I haven't tried STUN yet, but I read that that is generally unneeded.

i'm going to see if I can get the pap2t on a public IP address somehow to see if it's a router/firewall issue. I wish I had another IP phone to test with to rule out the pap2t.

Anyway, any insight would be much appreciated...

Thanks!

Update:
Looking through the logs I see this for the call that ended badly - sip.callwithus.com is ONLY used for outbound calling... the call in question was answered by the pap2t - so why is CWU hanging up?

code:

Oct 14 14:20:50 VERBOSE[6592] logger.c: -- SIP/cc-110-907e answered SIP/5068-081b57c8 Oct 14 14:20:50 VERBOSE[6592] logger.c: We're at 216.75.41.112 port 39122 Oct 14 14:20:50 VERBOSE[6592] logger.c: Video is at 216.75.41.112 port 44934  Oct 14 14:20:50 VERBOSE[6592] logger.c: Adding codec 0x4 (ulaw) to SDP Oct 14 14:20:50 VERBOSE[6592] logger.c: Adding codec 0x8 (alaw) to SDP Oct 14 14:20:50 VERBOSE[6592] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Oct 14 14:20:50 VERBOSE[6599] chan_sip.c: Hangup call SIP/CallWithUs-fd3a, SIP callid [EMAIL]18c220f00cfa2c0e703f27d26d7906ee@sip.callwithus.com[/EMAIL] Oct 14 14:24:00 VERBOSE[6592] chan_sip.c: Hangup call SIP/cc-110-907e, SIP callid 26d6cde41e76e83a4d1f83ba462a299d@216.75.41.112 Oct 14 14:24:00 VERBOSE[6592] chan_sip.c: Hangup call SIP/5068-081b57c8, SIP callid 22354b6b72aaaf95478806b30b19e6b1@204.11.192.27

RE: help - random hangups :(

Here are some thoughts and clarifications on the issue.

If two-way audio can be heard from both sides, for at least 5 minutes, it's not a regular NAT Traversal issue. Since the issue is not happening on a regular interval from the beginning of each call, I am almost willing to rule out a NAT Traversal issue completely. In addition the PAP2 doesn't support PPPoE, so I doubt it can acquire a true Public IP address.

Your trace indicates only the G.711 vocoders were used in the call. If CWU supports G.729, enable G.729 Passthru under General Settings, and place the G.729 as the Preferred Codec on the PAP2. Leave G.711 enabled otherwise the call will not connect via PBXes.

Ensure that DTMF is set to rfc2933 both on the PAP2 Extension and the CUW Trunk on your PBXes account. Set the DTMF Process AVT: Yes, the DTMF Process INFO: No, the DTMF Tx Method: AVT, the FAX Passthru Method: ReInvite, and the FAX Process NSE: No, on the PAP2 Line 1 or 2 configuration.

Try calling with this configuration and see if the issue you reported occurs. If it does, next in line is to setup a packet trace on your LAN.

RE: help - random hangups :(

Diafora,

You are truly amazing... I have not had a single dropped audio call in 2 days now!

The changes I made were:
* Set DTMF mode to rfc2833 on all trunks & extensions in pbxes
* DTMF Process INFO: No
* FAX Passthru Method: ReInvite (I think)
* FAX Process NSE: No
* All other suggestions I already had set

I would never have made those changes on the pap2t if you didn't suggest them. The only thing I didn't do was set g729 as the preferred codec.

Do you have any thoughts as to what change most likely was the 'fix'? The wife may be causing some talk-off on the pap2t - do you think it's safe to switch everything regarding DTMF to 'inband' - I have read that reduces talk-off - you agree?

Anyway, I just wanted to chime in and say thanks a million for those suggestions. I think I"m going to get an IP phone (A580ip maybe?) and kick the pap2t to the curb :/

Update: talkoff is pretty bad - especially when she says things like 'eye' or 'uh'. It appears to go in phases from a couple of times a minute to no problems...

RE: pap2t - help - random hangups :(

I am glad the above suggestions helped you improve the issue of dropped calls.

A couple of clarifications regarding DTMF transmission and vocoders. The InBand method requires the vocoder to be set to G.711a or G.711u. It transmits the DTMF tones as audible sounds during the call, similarly to the analog terrestrial PSTN telephony.

But when DTMF tones are transmitted via mobile phones, VoIP networks e.t.c, with no pristine signal conditions, the InBand transmission of DTMF should be avoided at all costs. This is precisely the reason d' être for the other DTMF transmission methods.

Voice is compressed and transmitted over somewhat unreliable networks where jitter and dropped packets are expected, therefore the newer DTMF transmission methods use alternative methods to transmit the tones.

Since G.711(a or u) is not as compact as G.729 or other lower rate vocoders, using it for calls outside of the LAN is not recommended. The MOS score difference between G.711 and G.729 vocoders, does not warrant hauling 5 times the packets across the Internet based on my tests.

Now the vocoder used is a purely personal choice. What I am somewhat hazy is on the definition of "talk-off". What exactly do you mean by that? What is actually happening on the call when this condition occurs?

Lastly regarding your option to upgrade from a PAP2 to a Siemens DECT IP-phone is a step in the right direction, since you will be eliminating a voice conversion step.

I haven't actually tried the A580 IP personally, but it shares code-base with the Gigasets I have. Starting with the 02184 version of firmware, the Siemens DECT IP-phones have become a lot more reliable, enough to be recommended for daily use. They are not perfect yet, but on their way there.

RE: pap2t - help - random hangups :(

Hi Diafora,

My understanding of 'talk off' is this:
When the pap2t is using an out of band DTMF like AVT, it has to analyze the voice stream and try to detect when someone presses a digit on the phone. If a persons voice matches the frequency of a DTMF tone, the pap2t wakes up and sends the tone across the internet causing a random beep on the other side of the conversation. There are lots of threads on dslreports about this and the pap2t in particular.

So, my question is: Of those changes I made, the switch from 'Auto' to 'rfc2833' probably wasn't the setting that fixed my dropped audio problem, right? I wish I knew what all those settings were for - such as 'FAX Process NSE'.

Thanks again.

RE: pap2t - help - random hangups :(

The most likely reason your calls were dropping off, based on your original description, is the detection of the Fax CNG Calling Tone during voice calls.

In the web interface of the PAP2 there are a couple of additional settings you can disable, to ensure no fax tones are processed in the middle of a voice call: FAX CED Detect Enable: No and FAX CNG Detect Enable: No

This way, no amount of "talk of" during a voice call, will trigger fax detection. The change you made earlier, confirmed the PAP2 was trying to detect fax tones during a voice call. If the above changes resolve the issue, you might not have to change the DTMF settings from RFC2833 to InBand.

Let me know how it goes.

RE: pap2t - help - random hangups :(

Hi Diafora,
Interesting post, am replying to it as it may just be related.
Got me wondering about a small problem we have always had. Occasionally when a pstn call comes in to the PBX via a SPA3102 FXO and is picked up by a PAP2 user there is a momentary very loud tone before the call proceeds normally. Users describe it as deafening. Could this be connected somehow with FAX operation, which we don't use by the way ?
Regards

RE: pap2t - help - random hangups :(

It might be closely related to FAX tone interception. As a rule of thumb, disable the FAX tones recognition on every ATA port and Inbound Route you don't expect a fax call. Just turn it on if and when you need it.

Simplify.

RE: pap2t - help - random hangups :(

Ok, on monday the problem presented itself again - loud dtmf done followed by one-way audio. Monday night I enabled the settings you recommended:
'FAX CED Detect Enable: No and FAX CNG Detect Enable: No'

And so far today (tuesday) has been good. I don't know why my wife is having such awful problems with this pap2t. I also changed all DTMF to inband to totally prevent the pap2t from processing anything - I hate this thing! We have a pretty stable DSL connection, so DTMF should be fine with inband.

I have placed an order for a A580IP phone for her and I will change the dtmf handling back to rfc2833 after I hook the phone up. Does 'auto' default to rfc2833?

Anyway, thanks for the help... I'll report back if the problem happens again with all the FAX detection settings disabled.

RE: pap2t - help - random hangups :(

When your A580 IP arrives, there will be a setting to enable RFC 2833 on the handset, and disable everything else.