[Freeswitch-users] Intermittent Audio Problem - Incomming Calls
Brian Campbell
bcxml at hotmail.com
Tue Jun 21 22:09:56 MSD 2011
Thanks David
I have followed your instructions and I find that the audio is definately flowing between FreeSwitch and Speech Server on calls that seem to be broken. So that points me to a Speech Server issue
Thanks so much for the advice, you have been a big help
Brian Campbell
From: david.ponzone at ipeva.fr
Date: Tue, 21 Jun 2011 16:41:14 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
Wireshark is your friend :)
Open your RTP flow, click on Telephony/RTP/Show All Streams
In the next window, select the stream you want (the one from FS to IVR) and click on Analyse.
And in the final window, click on Save Payload.
Then select the forward stream, and save as raw.
You should be able to import the raw file in any decent audio software (Audacity, etc...) as U-Law (I suppose you use PCMU), 8Khz.
David Ponzone Direction Technique
email: david.ponzone at ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 21/06/2011 à 15:13, Brian Campbell a écrit :
Thanks David
The Speech Server application reacts to speech from the caller validated by a grammar. There is a facility in Speech Server to log the calls, including the audio. On calls that work, I can hear the prompts from the application as well as my responses in the session audio. On calls that don't work, I hear the prompts from the application but none of my responses. After a small period of waiting the Speech Recognizer times out and tells me that it did not hear me say anything. I have tried increasing the timeout interval but the problem remains.
So I am not sure where the problem lies, is it something breaking down in Speech Server, or is the audio not making it from FreeSwitch to Speech Server.
One thing for sure is that it doesnt stop working half way through the call. It either recognizes utterances from the caller or it doesn't, which would lead me to think that sometimes the call is not properly setup between Speech Server and FreeSwitch, but as I said I have run out of ideas.
As I mentioned I do see RTP traffic, but I am afraid that I do not posses indepth knowledge about the protocol, so I am not sure how to properly analyse it.
Thanks again for your input
Brian Campbell
From: david.ponzone at ipeva.fr
Date: Tue, 21 Jun 2011 08:43:44 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
Brian,
twice you say "the IVR doesn't seem to hear anything the caller is saying".
I suppose that means it is an assumption because it does not react to DTMFs ?
Do you have the possibility in this IVR to record the call ?
This way, you would be sure.
Also you say that Wireshark tells you the RTP traffic is fine.
Even when the call is not working ?
Do you notice anything different in this RTP ?
You may also take a RTP trace between FS and IVR of a non-working call, analyse the RTP flows and save the payload as raw to a file.
You should be able to listen to the audio with the right tool.
David Ponzone Direction Technique
email: david.ponzone at ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 21/06/2011 à 04:29, Brian Campbell a écrit :
I have run into a very strange issue with periodically losing Audio on Incomming calls
I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
<extension name="Inbound_FromSoTel">
<condition field="destination_number" expression="^1(\d+)$">
<action application="set" data="sip_h_X-FS_UUID=${uuid}" />
<action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" />
</condition>
</extension>
Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong.
Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
Can anyone advise on what the issue might be. I am completely out of ideas.
Thanks
Brian Campbell
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110621/6ffb5f62/attachment-0001.html
More information about the FreeSWITCH-users
mailing list