Audio delay when answering calls

Forums › VoIP › Audio delay when answering calls

BobTBunny

15 posts

Geek

#111622 9-Nov-2012 12:35

Hi all,

We have been having an intermittent issue on our VoIP phone system since it was installed. I'd rather not mention the brand initially, as it sounds like we're the only customer of theirs in the world that has this issue. I.e. I don't want to tarnish their (I assume) othwise good reputation.

The problem we have is when you answer a call, there can be a delay in audio of anywhere from 2 seconds, up to over 30 seconds. Neither party can hear the other, it's dead silence on the line. There's also no audio loopback; you can't even hear yourself talking to them.

The call could be answered by Group Pickup, transferred to your extension by someone else who had answered the call, or even just an internal call between extensions in the office.

It happens to at least 6 of our 24 extensions. I say at least because some staff don't get many calls, so they may be having smaller delays and not noticing.

I have swapped network cabling, the PoE switch, phisically segregated the phone system from the rest of the network (not just made VLans), swapped some phones for new ones, even replaced the main phone system boxes that "do the work".

If anyone has any suggestions of other things to try I'd greatly appreciate it.

Thanks,

Bob

1 | 2

3344 posts

Uber Geek

Trusted

Vocus

#714640 9-Nov-2012 13:07

Check any settings relating to "early media" on those extensions.

Also; hows latency between the server/PBX and those extension?

Sharesies

Trade NZ and US shares and funds with Sharesies (affiliate link).

sbiddle

30853 posts

Uber Geek

Retired Mod

Trusted

Biddle Corp

Lifetime subscriber

#714645 9-Nov-2012 13:10

Are all the phones on the same network segment as the PBX? Is there a firewall / router between the phones and PBX?

joshp

205 posts

Master Geek

Trusted

WorldxChange

#714785 9-Nov-2012 16:03

Is the phone system set up to act as the media bridge? It could be that the phones are trying to do the media directly with each other and are running into issues. You should see this in the INVITEs/200 OKs SDP. You may even want to try disabling the phone system as acting as a bridge.

I would at least try and get a switch configured up with a mirror port that is outputting to a laptop with wireshark on it (hubs are a little harder to come by these days), this way you can see in real time if the audio is actually being sent from the PABX to the phone and vice versa. Try and track the problem down that way.

Also is there a certain volume of calls that you get before you notice the issue?

Cheers

Josh

http://wxc.co.nz

BobTBunny

15 posts

Geek

#714791 9-Nov-2012 16:16

Thanks for the responses. I'm going to answer both at once.

The phones and system units are all plugged directly into the PoE switch. No firewalls between. The switch is basically standalone, with just the phones, system PBX units and a cable directly to our Cisco 1940 router. No connections to other switches or anything fancy.

Latency is minimal, consistent <1ms (using a pc plugged into the back of the phone)

All the devices are on the same floor of the building, computers don't show network problems using the same cabling. The phones can only connect at 100mbps, so they're not trying to run gigabit over cat5 cabling. I've also plugged some into new cat6 cabling that was fresh with the phone system install (same story as every other company's network cabling - we didn't have enough sockets) that pc's happily run gigabit over. so I'm confident the network cabling is good.

ubergeeknz: I can't see any "early media" settings in the phone system. Could it go by another name? From what I've just read, it seems to be primarily for PSTN calls, but this happens on internal extension-to-extension calls as well. Presumably (and please correct me if I'm wrong) with the phone extensions being configured the same, they would both either support it or not, either way it shouldn't be a problem?

If it sounds like I don't know much about SIP, it's because I'm not the phone system implementer. I'm just the network admin looking for answers while the phone system people try to get answers from their supplier and the manufacturer for a problem that's been going on for >6 months. In other words, please don't shoot me down as yet another person trying to install SIP when they don't know what they're doing. It's not me that did it :-]

Thanks,
Bob

ubergeeknz

3344 posts

Uber Geek

Trusted

Vocus

#714796 9-Nov-2012 16:20

BobTBunny:
ubergeeknz: I can't see any "early media" settings in the phone system. Could it go by another name? From what I've just read, it seems to be primarily for PSTN calls, but this happens on internal extension-to-extension calls as well. Presumably (and please correct me if I'm wrong) with the phone extensions being configured the same, they would both either support it or not, either way it shouldn't be a problem?

Sometimes it's called "Delayed Media" and works in reverse... it could also be called any number of other things. But with LAN latency it shouldn't be an issue anyway.

ubergeeknz

3344 posts

Uber Geek

Trusted

Vocus

#714833 9-Nov-2012 17:18

Are you able to get a network capture at the phone endpoint that's experiencing delayed audio, and one at the PABX/Server (simultaneously) This would almost certainly narrow down the issue.

BobTBunny

15 posts

Geek

#715021 10-Nov-2012 07:49

JoshP: I'll answer this easy bit first. It seems to happen with any amount of load on the system, from this being the only call, up to if we have 10 other people on calls.

As for the packet sniffing, we're waiting for the distributor to tell the installers what they need to trace, then we'll get WireShark set up. I'm trying not to get into that bit myself, I'd rather make the people that sold it to us do the really hard work.

Bob

bender

220 posts

Master Geek

#718591 17-Nov-2012 07:15

Can you try making a call out, and start talking as soon as you place the call? See if you hear the other end immediately.

Eg start counting to 20

kornflake

409 posts

Ultimate Geek

#718737 17-Nov-2012 12:46

what pbx software are you running

BobTBunny

15 posts

Geek

#719853 20-Nov-2012 00:03

bender: Can you try making a call out, and start talking as soon as you place the call? See if you hear the other end immediately.

Eg start counting to 20

It's incoming calls that sometimes have the delay, not outgoing. Outgoing is fine.

The exception to that is once when it was a purely internal call, from one extension to the other in the same building. So it can't be an issue with the SIP provider.

The delay is seemingly random, from 1/2 a second (I know, that's really nothing in the grand scheme of things, but normally the audio is "right there" immediately) all the way up to 35 seconds. We have customers hanging up because they think they're having phone problems.

It happens on random extensions, and can be either on a "group pickup" or a transferred call to you.

Of the hundred or so calls we get a day, anywhere from 3 - 10 have a delay on a given day, that I get told about (or experience personally).

Cheers,

Bob

BobTBunny

15 posts

Geek

#719855 20-Nov-2012 00:06

kornflake: what pbx software are you running

It's a hardware system, not software. If the installers can't find the problem by the end of the week, I'll name the brand. It's not (what I'd call) a well known brand when it comes to SIP phone systems. I didn't know they made them.

So you can take it from that it's not Cisco, Avaya, snom, etc.

Bob

maverick

3594 posts

Uber Geek

Trusted

WorldxChange

#719877 20-Nov-2012 06:53

Interesting ... I actually like faults like this even though they are very very frustrating.

You assumptions will be correct in that it is not your Sip provider as you experience the issue internally, your only option really is to packet trace one of these calls and sine you can reproduce it it should not be to hard to find.

Do you have a support contract with the supplier of the system ?
Does your system support logging or SIP tracing ?

If it doesn't then what I would do is mirror port 2 phone and make run a trace whilst you make a call between the 2 phones , restart the trace between calls so you have a full trace for each call, if you can reproduce the issue leave the call up till the audio starts then stop the trace, make sure you only mirror the 2 test phones so you dont get any other traffic, if you can capture the issue it will be very easy to identify the issue, happy to take a look at the trace for you as well, as I said I actually like these types of faults.

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

https://www.facebook.com/wxccommunications

theEd

341 posts

Ultimate Geek

Trusted

#719925 20-Nov-2012 09:13

All I can think of is that its trying to establish the RTP stream via one method, failing, and falling back to another. But that should be very obvious to the installer once they do some SIP traces.

If it eludes them, ask for a copy of the pcap files and I'm sure myself or one of the other SIP guys here can take a look

kornflake

409 posts

Ultimate Geek

#720333 20-Nov-2012 20:22

please dont say vadacom, my guess is epigy

kornflake

409 posts

Ultimate Geek

#720360 20-Nov-2012 20:52

will take the vadacom statement back. to be fair the mate was only giving us one side of the story, guess what in the end

........ he was wrong

1 | 2