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lifespeed

join:2009-09-08

[Voip.ms] incoming call took 10 seconds for outgoing audio

What an awkward moment. I received a call on voip.ms, Asterisk 1.8.3, and it took 10 seconds before the caller could hear me say "hello".

This happened twice today, and appears to be a new problem on the same VoIP setup I have been using for two years.

I use losangeles.voip.ms. I'll try switching servers and report back. I have had some other minor glitches with voip.ms in the past, including a slight delay to establish audio. But this time is the worst as far as duration

I don't have many RTP audio ports open in asterisk, as I try to prioritize incoming RTP on my network (which has no QoS tags). Opening fewer ports makes it more likely the high-priority range will actually be VoIP traffic.

Long story short, I only have 100 ports open, but just increased that to 2000 and see if it makes a difference.
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Lifespeed


VexorgTR

join:2012-08-27
Sheffield Lake, OH
kudos:1
Yowser.... VoipMS has been having a bad week based on all the posts lately. Best wishes on getting things back running.

lifespeed

join:2009-09-08
reply to lifespeed
I opened the additional RTP ports and made a test call into my system (still using losangeles.voip.ms) with two different mobile phones.

Sample size of two, it was very fast response. I don't think that is enough to declare victory, but I'll update status here.

My understanding is RTP audio isn't established until the call is answered. So if there is any delay it will be heard. Either that, or I have no idea what I am talking about.
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Lifespeed

nitzan
Premium,VIP
join:2008-02-27
kudos:8
reply to lifespeed
Are you behind a NAT? do you have STUN enabled on Asterisk or your adapter/phone? if yes- disable it.

lifespeed

join:2009-09-08
STUN has always been disabled.
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Lifespeed

advocate99

join:2011-03-08
reply to lifespeed
Try another server and see if it still happens.
Try another DID (preferably from another area) and see if it happens.
Try another provider (Callcentric has free phone numbers) and see if it happens.

Use the results above to deduce the cause and open a trouble ticket with the responsible party.

kaila

join:2000-10-11
Lincolnshire, IL
reply to lifespeed
said by lifespeed:

....Sample size of two, it was very fast response. I don't think that is enough to declare victory, but I'll update status here.

My understanding is RTP audio isn't established until the call is answered. So if there is any delay it will be heard. Either that, or I have no idea what I am talking about.

Hope it stays resolved. I have to say the vast majority of these types of issues I've had or clients have had, reside within the local network.

SIP packets between the source and destination set up the call, and contain information about what codecs the sender supports and what RTP port the receiving end should open to establish the audio.


crazyk4952
Premium
join:2002-02-04
united state
kudos:1
reply to lifespeed
You are not the only one having this issue lately. See: »[Voip.ms] Delay in Answering Calls

lifespeed

join:2009-09-08
reply to kaila
said by kaila:

Hope it stays resolved. I have to say the vast majority of these types of issues I've had or clients have had, reside within the local network.

SIP packets between the source and destination set up the call, and contain information about what codecs the sender supports and what RTP port the receiving end should open to establish the audio.

I hope so too. While I understand that there are plenty of opportunities for a user to break or degrade VoIP based on network configuration, I will point out that my setup has been working for a looooong time. And I would like to think I am somewhat familiar with how this works by now.

One change at a time. I'll observe how the increased number of forwarded and high-priority (QoS) ports for RTP audio in Asterisk and my router work and report back with results over the next few days.
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Lifespeed

kaila

join:2000-10-11
Lincolnshire, IL
said by lifespeed:

I hope so too. While I understand that there are plenty of opportunities for a user to break or degrade VoIP based on network configuration, I will point out that my setup has been working for a looooong time....

Don't rule out hardware issues. As stuff ages anything can happen causing intermittent problems which can be absolutely maddening to track down. Glitchy memory, boards, sagging power supplies, cables, physical ports, literally anything goes- and will go eventually.
Expand your moderator at work

lifespeed

join:2009-09-08

1 edit

1 recommendation

reply to lifespeed

Re: [Voip.ms] incoming call took 10 seconds for outgoing audio

Today I received a phone call. The caller did not hear ringback, and did not hear my voice for a good 5 - 10 seconds after I answered.

I am more than willing to examine my setup, but find it highly suspicious that a working setup suddenly has this problem. Not to mention it seems to be a common problem for others.

Edit: I submitted a ticket to voip.ms

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Lifespeed

lifespeed

join:2009-09-08
reply to lifespeed

no long delays so far, but ringback (early media) disappeared

I switched to seattle server, and after a few days I am not seeing the long delay before audio goes through. But I have also lost the ringback tone (early media). I can dial an early media test number, and it works on my setup.

Really not stellar results. I should be able to use losangeles server for the shorter ping, and I should be able to receive early media. I guess the audio delay is the most irritating problem so for now I'll stay on seattle.
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Lifespeed

lifespeed

join:2009-09-08
reply to lifespeed

Re: [Voip.ms] incoming call took 10 seconds for outgoing audio

Just switched back to losangeles server to see if it still behaves badly with delayed connection of audio. Hopefully ringback functions consistently.