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Summary:ASTERISK-03353: Audio Packets Being Sent Prior to IAX ANSWER
Reporter:velochap (velochap)Labels:
Date Opened:2005-01-25 19:34:08.000-0600Date Closed:2011-06-07 14:10:10
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) iaxdump.txt
( 1) iax-inbound-good
( 2) iax-outbound-bad
Description:The high level symptom is that when placing a call via IAX the called party doesn't hear the caller for the first 2-3 seconds.  This results in "Hello, Hello.  Are you there."

Upon closer inspection it appears that the bearer/audio channel is being cut through before the call has completely finished setting up.   This can be viewed in the attached debug where you can clearly see that there is approx 3 seconds of ulaw audio present before the ANSWER packet is being received.  This is obviously causing the missed audio as * is most likely not cutting through the audio until the ANSWER is received.
Comments:By: velochap (velochap) 2005-01-25 19:35:42.000-0600

Sorry should be a MINOR bug...obviously not FEATURE.
By: Mark Spencer (markster) 2005-01-25 20:30:11.000-0600

Audio is supposed to be cut through immediately.  It's not supposed to wait until there is an answer.
By: Mark Spencer (markster) 2005-01-25 20:30:24.000-0600

What is the client / setup at each end in any case?
By: velochap (velochap) 2005-01-25 21:31:23.000-0600

I have tried a good handful of providers to make sure that the problem isn't with the provider.  The IAX providers that I have tried and experience this symptom are VoicePulse, IAX.cc, TELIAX and VoIPJet.   The problem remains regardless of where I point the traffic.

I decided to submit a bug as I noticed that a number of others are also experiencing this same symptom as posted on asterisk-users yesterday.

The current setup is as follows:

SIP client----*----FW--------iax.cc

My iax.conf file is below.

I also verified that the load on the server is almost zero.  xenon server running asterisk as a dedicated box with a single call.   OS is CentOS 3.

Also, once the call is setup the audio is perfect aside from approx the first 3 seconds of the call where there is no audio.

Aside from the IAX debug and IAX.conf is there any additional information that I can provide which would isolate the problem?

[general]
port=4569
bindaddr=10.1.1.17
bandwidth=high
disallow=all
allow=ulaw
trunk=no
tos=184
jitterbuffer=yes
dropcount=1
maxjitterbuffer=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1


register => username:pass@iax2.sixtel.net

[sixTel]
type = friend
host = iax2.sixtel.net
context = in-out
secret = pass
disallow = all
allow = ulaw
By: Mark Spencer (markster) 2005-01-26 00:18:29.000-0600

Can you test with a zap device instead of a  sip phone and see if the problem still remains?
By: velochap (velochap) 2005-01-26 07:19:43.000-0600

Another poster on asterisk-users did say that they tried this from a zaptel and the problem still remained.  Another also pointed out that it exists with both SIP and IAX calls.

Here's the thread:
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/77378
By: reidfo (reidfo) 2005-01-26 08:38:35.000-0600

I posted my results with this to the list a couple of days ago.

Here are my notes:
1) I experience this problem with both SIP and IAX outbound calls, on nearly every call.
2) I have tried using G.711 ulaw, GSM, iLBC all with the same results.
3) Doesn't matter if call is from a SIP phone or Zap FXS channel
4) Occurs with multiple SIP/IAX providers
5) Has occurred on every version of * I have tried over the past year
6) Occurs on every * install I have performed
7) Using Mandrake 10.0 and 10.1
8) * server has 0 load on it
By: velochap (velochap) 2005-01-26 08:49:51.000-0600

I would also add that this is occuring on my end also for *outbound* calls only.  Inbound calls do not experience this symptom.  As I pointed out previously, I initially thought that it must be the provider.  So I have tried at least 4-5 providers now and the problem follows.  

I am going to get an ethereal capture of the traffic during one of these calls and post it.  Hopefully this will lend some insight into the problem.
By: velochap (velochap) 2005-01-26 10:06:37.000-0600

Ok, I have added two capture files "iax-inbound-good" and "iax-outbound-bad".  Both files are in libpcap and can be viewed in ethereal.  

The first file is an inbound IAX call which does not exhibit the condition where the first three second of the audio is cutoff.  All inbound calls do no exhibit this condition.

The second file is an outbound IAX call which does exhbit the condition of the first three seconds of audio being lost.  All outbound calls experience this regardless of the provider.

The devices in the files are as follows:

10.1.1.5 = SIP Device which is initiating and terminating the call

10.1.1.17 = local * server

205.234.133.203 = remote * server managed by iax.cc (in this example, again same problem is shown with other providers)
By: velochap (velochap) 2005-01-26 10:20:11.000-0600

So after closely examining the good and bad packet captures I am observing that good calls have a IAX ANSWER packet before any audio is sent.   The bad calls don't see an ANSWER packet until well after (seconds) the audio is being sent.  

In the bad packet capture, notice that the ANSWER doesn't come until packet # 1086.   The good capture doesn't have any audio packets prior to the answer packet.

I understand that the protocol was designed to have the audio cut through immediately.  Is it possible that something else is broken that isn't forwarding the audio until the ANSWER is seen?   This is the only variance that I see between sucessful and unsucessful calls.
By: reidfo (reidfo) 2005-01-26 18:09:14.000-0600

Once again, I'd like to stress that this behavior is not limited to IAX. It happens with SIP outbound calls as well.
By: Clod Patry (junky) 2005-02-11 19:05:57.000-0600

Where are we with this bug ?
any updates?
By: Kevin P. Fleming (kpfleming) 2005-02-12 10:15:01.000-0600

My users have occasionally reported this problem as well, but it's very rare in our case. I think it warrants looking into, as when it happens it is very annoying :-)
By: zoa (zoa) 2005-02-12 12:19:43.000-0600

maybe a weird question, but are you using a jitter buffer ?
By: Kevin P. Fleming (kpfleming) 2005-02-12 12:29:38.000-0600

In my case, no. And come to think of it, I have not had any reports of this problem since we stopped using IAX and switched over to SIP between our servers.
By: reidfo (reidfo) 2005-02-12 13:26:07.000-0600

I've tried both with and without jitter buffer. Same results each time.
By: Mark Spencer (markster) 2005-02-12 13:32:06.000-0600

What appears to be the problem when you look at the trace?
By: reidfo (reidfo) 2005-02-12 14:05:13.000-0600

I can't tell there's a problem from the trace, now that I know audio is supposed to cut through before answer. All I can tell is that the symptom is the first 1-4 seconds of inbound audio on an outbound SIP or IAX call are cut off.

It does not appear that audio is muted during this period. I hear ringback tone, and then audio immediately follows the ringback. Problem is that the audio coming thorough is 1-4 seconds into the call.
By: Mark Spencer (markster) 2005-02-27 21:09:01.000-0600

Between frame 544 and frame 779, there is silence receivecd on the IAX side for over two seconds.  This would appear to be why there is a missing piece of conversation, although it clearly takes place before the answer by some 2 seconds.
By: Mark Spencer (markster) 2005-03-10 02:59:46.000-0600

Closing due to lack of response from bug placer.