Re: Problem with asterisk

[ronaldo]

Hi all,
Here is my problem:
This appear when i made a video call with very hight bandwith(3Mbps).

Audio call is have no any problem, but bandwith still a prolem there.

 == Using SIP RTP CoS mark 5
    -- Executing [800@default:1] Answer("SIP/1060-00000000", "") in new stack
[Sep 26 07:45:36] WARNING[3942][C-00000000]: channel.c:1307 __ast_queue_frame: Exceptionally long queue length queuing to SIP/1060-00000000
[Sep 26 07:45:36] WARNING[3942][C-00000000]: channel.c:1307 __ast_queue_frame: Exceptionally long queue length queuing to SIP/1060-00000000
[Sep 26 07:45:36] WARNING[3942][C-00000000]: channel.c:1307 __ast_queue_frame: Exceptionally long queue length queuing to SIP/1060-00000000
[Sep 26 07:45:36] WARNING[3942][C-00000000]: channel.c:1307 __ast_queue_frame: Exceptionally long queue length queuing to SIP/1060-00000000
    -- Executing [800@default:2] Playback("SIP/1060-00000000", "beep") in new stack
    -- <SIP/1060-00000000> Playing 'beep.gsm' (language 'en')
    -- Executing [800@default:3] Wait("SIP/1060-00000000", "1") in new stack
  == Spawn extension (default, 800, 3) exited non-zero on 'SIP/1060-00000000'

[Anton]

Hey ronaldo, if you have no audio problem, would you please tell something about topic original question?

[Hadi Ams]

Hi Anton ,

I have the similar problem when I connect from chrome on Ubuntu 12.04 (same pc I am running Asterisk on ).

incoming calls are ok , but calls from chrome are the same as your problem. I see same errors in Asterisk console.

wireshark shows packets are being transmitted between chrome and asterisk but no packet is being sent from asterisk to the 2nd client ( xlite or any other sip , in my case a pstn gateway ).

I am hearing either noise or silence in chrome.

but running chrome and xlite on another pc works fine for me and I have two way audio for incoming and outgoing.

are you using chrome on the same pc that you have Asterisk ? try to use another pc just to make sure your configuration is ok.

This problem has been reported several times.

I'm looking forward to any help from experts ...

Thank you .

[ronaldo]

hi, about my console and video prolem.

Vào 12:38:22 UTC+7 Thứ tư, ngày 26 tháng chín năm 2012, Anton đã viết:

[Anton]

Hi Hadi,
Thank for your reply, it encouraged me a bit :)
In my case I use one vm (vm1 debian 6) with asterisk, another one (vm2 also debian 6) with apache and sipml5. These vm's are deployed on my pc(win7) with vmware server. I'm running chrome pc with win7. After your reply I tryed to run chrome on vm2 and got the same result. So, these pc's are different (but may be something wrong with virtualization). I also run wireshark on machine with asterisk and really there are packets from chrome to asterisk, asterisk to chrome, softphone to asterisk and no packets from asterisk to softphone.

среда, 26 сентября 2012 г., 15:21:27 UTC+6 пользователь Hadi Ams написал:

[Mamadou DIOP]

If you can share the full trace by opening a new ticket in the issue tracker we could help. Also provide information on your chrome version.

If you're not making changes in the sipml5 you don't need to host it on your Apache. You can directly use "sipml5.org/call.htm" to be sure to always use the latest version.

--
 
 

[Anton]

Hi, Mamadou, thank you for reply.
I use chrome 21.0.1180.89
Ok, I'll create an issue. I host it in own apache from earlier versions, because of hardcoded addresses (now i reverted all my edits, thanks for expert mode page) and I updated it today. Also it's not possible for me to use sipml5.org, because of firewall and lan. Please give me direction on full trace, do you mean wireshark trace  or chrome log or chrome console? (and what call direction chrome=>softphone, softhpohone=>chrome or both)
I can share :)

[Mamadou DIOP]

chrome console (attached as *.txt) from the starting of the browser to call ending.

"because of firewall and lan" -> lan should not be an issue (thanks to ICE). For firewall do you mean UDP ports are blocked?

--
 
 

[Anton]

For firewall I mean that my asterisk in lan behind firewall and I can't setup port forwarding for it, so, I can't use sipml5.org/call.html with this asterisk.
I've submitted issue 44 as you ask  https://code.google.com/p/sipml5/issues/detail?id=44
Thank you a lot.

[Mamadou]

To be sure that there is no Firewall issues try to create two sip
accounts on any public SIP service (e.g. one from
http://code.google.com/p/sipml5/wiki/Public_SIP_Servers) and try calls
between two chromes

[Hadi Ams]

When I connect using chrome on ubuntu-64 bit ,in outgoing calls I hear noise and as I explained in issue 44 , the problem is with the crypto attributes in outgoing SDP and I have similar issue when I use my own Web and Asterisk server.

Incoming calls are always ok , because Asterisk always offers : a=crypto:0 AES_CM_128_HMAC_SHA1_80 which is the correct one .

Using chrome in win7-32 for outgoing and incoming works like a charm ( both with online server and my own ).

You mentioned somewhere that even with online servers you are hearing noise when you call from chrome xp to chrome mac.

Can you please verify what you are seeing in outgoing INVITE/SDP ?

you should probably see something like this :

a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:FH1ALT1PzeMmcat54E72QOmWIcXzlyPZMtLfMyq4

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:flRFtbPtSR9wheo4pmzy3R2bc7rewtLdWN0OCROI

which after negotiation only AES_CM_128_HMAC_SHA1_32 survives and causes the trouble.

Thanks

On Thursday, September 27, 2012 5:58:39 PM UTC+2, James Mortensen wrote:

Hi Hadi,

If it helps, the errors and warnings you report in Issue 44 are the same ones I'm getting.  However, I don't seem to have any problems connecting to the sipml5.org server that will be up until Friday. Do you see your same issues when connected to 1060 and 1062 at ws://sipml5.org:8088/ws or do things work well for you there too?

James

[Hadi Ams]

There are some good news at issue tracker page : http://code.google.com/p/sipml5/issues/detail?id=44

Looks like the issue is fixed now .(or at least my problem is solved now.)

regards

Hadi