|Summary:||ASTERISK-01935: Poort Quality Issues using IAX2 and Cisco 7960 SIP Phones|
|Date Opened:||2004-07-01 09:44:56||Date Closed:||2011-06-07 14:04:50|
|Description:||I'm able to reproduce this quite easily. I have a test Asterisk Server (we'll call it B), and I have it register with voicepulse via IAX. I then have my Cisco 7960 connecting to my Asterisk server via SIP. Whenever I receive calls, VoicePulse sends it into Asterisk, the phone rings, but the caller (the remote side) breaks up. It usually occurs within a minute. They start to sound choppy, poor quality, like a cell phone losing signal. I also hear repeated words a lot, they say something and I hear it twice. I then tried to connect Asterisk to NuFone via IAX and the same problem occurs. When I place outbound calls, through NuFone or VoicePulse, the person I'm calling (again the remote end) sounds choppy and distorted, the remote end always seems to hear me fine though. |
For sometime now we've been using a main asterisk server, I'll call it "A", for the entire office of about 50 people. It pushes all it's outbound calls to a Cisco 3640 with 4 PRI's in it. This system has worked great for some time now. I made my Asterisk "B" server connect to my Asterisk
A" server via IAX. Whenever I call (using my Cisco 7960 connected to my B asterisk server) the "B" Asterisk pushes it to "A", and "A" pushes it out the Cisco Router. Again, quality breaks up on the remote side. If they call in, and I have Asterisk Server "A" push it to "B" and it rings my phone, remote end has quality issues. If I make my Cisco phone connect directly to Asterisk Server "A" and have it call out, it gets pushed to the Cisco router and everything sounds crystal clear on both ends.
I've tried every option I can think of in iax.conf, jitterbuffer, tos, different codecs for both the phone sip channel and the iax channel, iaxcompat, bandwidth=, etc. to no avail. The interesting thing is, my Sipura SPA-2000 doesn't seem to be affected by this problem. Quality is normal. I also tried a Cisco ATA 186 and it too seems immune. I thought it was just Cisco phones, but then I tried it with X-Lite and it saw the problem as well. I've tried various cisco firmwares for the phones, from 3-6-2 up to 3-7-1 the latest. No luck. Different firmwares seem to be less affected by the problem, but it's always slightly noticeable. I've even had it where two phones with the same firmware, 1 see's the problem, the other does not. It's all on an internal network so there shouldn't be congestion issues. I'm really at my wits end, at a complete loss. Our sister/partner company is also messing with Asterisk and I've been talking with their Asterisk tech who's just learning but learning fast, and he's able to recreate the problem as well.
****** ADDITIONAL INFORMATION ******
I've reproduced this on Asterisk CVS 6-15-04, 6-16-04, 6-22-04, 6-24-04, 6-27-04 and 6-29-04. I've used different server hardware with different Linux distro's on 2 completely different networks (DS3 at work, and my 1200/1200 SDSL at home). They all see the issue. VoicePulse/NuFone/My PRI's. Calls from phone to phone are great, phone to phone through 2 different asterisk's connected via IAX also are fine. It's just when we hit the PSTN. Is it a jitter issue? Cisco 7960? IAX conversion issue? Asterisk?
|Comments:||By: zoa (zoa) 2004-07-01 11:51:32|
hmmz, i'm using cisco 7960 and 7905 without any problems...
But i'm still on CVS-HEAD-05/14/04-20:53:48
By: webunited (webunited) 2004-07-01 11:59:51
I have a feeling it's been introduced as of late. I spoke with a tech at NuFone who confirmed that users with 7960's and Asterisk have reported problems connecting through his system. He thought it was a Cisco bug, but after some testing we discovered this might not be so.
Anyways, I just downgraded to 0.9.1 and the problem APPEARS to have gone away. All quality is fine, pushing out through VoicePulse right now. I've tested by having somebody calling my test numbers all over the U.S and I can hear them fine. I then called them back, through voicepulse/iax and I could still hear them fine.
I then tried 3 different Cisco phones, firmware P0S3-5-3-00 P0S3-6-3-00 and P0S3-7-1-00 and before they'd all have difference of opinions on which call would be bad and which would be good. Now they're all consistant, they all sound good. There is a slight concern, when they first connect on the call, I hear a slight distortion, but then it goes away. Could be normal. But it scares me more distortion could return.
I think it may be a new bug somehow introduced in the CVS versions sometime prior to 0.9.1; Can you backup your /var/lib/asterisk/ /var/spool/asterisk/ /etc/asterisk/ and /usr/lib/asterisk/ directories and then try upgrading to the latest CVS to see if the problem occurs, same config and everything?
I'm now going to re-upgrade to the 7-1-04 CVS and then overwrite my chan_iax2.so file with the one from 0.9.1 to see what happens.
By: webunited (webunited) 2004-07-01 14:03:00
Scratch that idea. Tried to use the latest CVS asterisk, with the chan_iax2.so module from 0.9.1 but got this:
Jul 1 12:57:47 WARNING: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_manager_register
Jul 1 12:57:47 WARNING: Loading module chan_iax2.so failed!
The problem is strange though. I went back down to 0.9.1 and the IAX problem seems to be 99% better. But music on hold got broken somehow, guess my configs are too new or something, so I went back to the CVS 7-1-04 and did a make upgrade. The music on hold problem went away, but my IAX2 outbound calls to VoicePulse were still problematic. However, they seem to be less problematic, like 90% of the problem was fixed. I then went to the CVS-7-1-04/asterisk/ directory and did a make install. Now the problem is back full force and worst than ever. It's probably coincidental but right I'm just tired of this problem. I'm stuck. I don't think I can go back to 0.9.1, not until I figure out which options is breaking music on hold (maybe I have to also downgrade libpri/zaptel). I'd prefer the latest CVS as there are a lot of other bug fixes. But I can't deal w/ whatever's breaking IAX to SIP translation in conjunction with a Cisco 7960.
By: twisted (twisted) 2004-07-01 16:19:21
Just to add to this, as I have experienced bad sound quality from my 7960 and VoicePulse, it appears that voicepulse is sending bad timestamps, which, in fact, causes the bad audio on the 7960's. However, I am also a NuFone customer, and have had no problems with calls on my 7960 from NuFone, or any other IAX devices.
This appears to be a voicepulse issue, we really should find out what version of * they are using on their servers, as the timestamp slip bug may not have been fixed on their end.
By: Mark Spencer (markster) 2004-07-01 18:16:41
I'm almost positive this is a question of what version of asterisk voicepulse is running. You can use ethereal to confirm the timestamps are incorrect.
They differ only by about 1ms, but for whatever reason on the Cisco phone this causes extreme audio issues.
By: Brian West (bkw918) 2004-07-01 19:14:38
I can confirm that 7960's with 7.1 do not have this issue via NuFone. This is an issue with the version voicepulse is using.
By: Brian West (bkw918) 2004-07-01 22:51:07
This isn't a bug in cvs-head.. voicepulse needs to update.