[sip-comm-dev] Re: One-way Audio Problem


#1

Hi Damian,

Thanks for tracking this problem.
You will be interested to know that I tested the asterisk server with two X-lite 3.0 clients and there was no audio problem.
Next, I used the sip-communicator client and an x-lie 3.0 client:The 2-way communication worked when the sip-communicator client initiated the call to the x-lite client. However, the audio problem was observed when the call was initiated by x-lite client to the sip-communicator client.

Please let me know you thoughts on the above

Gus

···

-------------

Date: Mon, 11 Jun 2007 10:18:30 +0300

From: Damian Minkov <damencho@damencho.com>

Content-Type: text/plain; charset=windows-1252; format=flowed
Subject: [sip-comm-dev] One-Way Audio Problem

Hi,

I have been tracking this last week and the problem is that asterisk
doesn't send the last ACK message to sip-communicator thats why it stays
in state
"connecting".

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#2

Hi Gus,

I'm facing the same problem, did you resolve it?? How can I make asterisk
send the ACK to the calee?? It's quite strange because I was using the same
asterisk configuration and all was perfect! I was able to establish a
conversation between two SC clients but since few days the caller status
changes to "connected" while the callee status stays "connecting".

Thanks a lot for help.

···

On 6/11/07, Gus Samba <gus_samba@hotmail.com> wrote:

Hi Damian,

Thanks for tracking this problem.
You will be interested to know that I tested the asterisk server with two
X-lite 3.0 clients and there was no audio problem.
Next, I used the sip-communicator client and an x-lie 3.0 client:The 2-way
communication worked when the sip-communicator client initiated the call
to
the x-lite client. However, the audio problem was observed when the call
was initiated by x-lite client to the sip-communicator client.

Please let me know you thoughts on the above

Gus
-------------

Date: Mon, 11 Jun 2007 10:18:30 +0300
From: Damian Minkov <damencho@damencho.com>
Content-Type: text/plain; charset=windows-1252; format=flowed
Subject: [sip-comm-dev] One-Way Audio Problem

Hi,

I have been tracking this last week and the problem is that asterisk
doesn't send the last ACK message to sip-communicator thats why it stays
in state
"connecting".

_________________________________________________________________
Get a preview of Live Earth, the hottest event this summer - only on MSN
http://liveearth.msn.com?source=msntaglineliveearthhm

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#3

Hi,

can you give some more info. as is the callee behind NAT or not? Can you send me a dump of the sip traffic between asterisk and sip-communicator. Which version of sip-communicator and asterisk are you using?

damencho

asmouta wrote:

···

Hi Gus,

I'm facing the same problem, did you resolve it?? How can I make asterisk send the ACK to the calee?? It's quite strange because I was using the same asterisk configuration and all was perfect! I was able to establish a conversation between two SC clients but since few days the caller status changes to "connected" while the callee status stays "connecting".

Thanks a lot for help.

On 6/11/07, *Gus Samba* <gus_samba@hotmail.com > <mailto:gus_samba@hotmail.com>> wrote:

    Hi Damian,

    Thanks for tracking this problem.
    You will be interested to know that I tested the asterisk server
    with two
    X-lite 3.0 clients and there was no audio problem.
    Next, I used the sip-communicator client and an x-lie 3.0
    client:The 2-way
    communication worked when the sip-communicator client initiated
    the call to
    the x-lite client. However, the audio problem was observed when
    the call
    was initiated by x-lite client to the sip-communicator client.

    Please let me know you thoughts on the above

    Gus
    -------------

    Date: Mon, 11 Jun 2007 10:18:30 +0300
    From: Damian Minkov <damencho@damencho.com
    <mailto:damencho@damencho.com>>
    Content-Type: text/plain; charset=windows-1252; format=flowed
    Subject: [sip-comm-dev] One-Way Audio Problem

    Hi,

    I have been tracking this last week and the problem is that asterisk
    doesn't send the last ACK message to sip-communicator thats why it
    stays
    in state
    "connecting".

    _________________________________________________________________
    Get a preview of Live Earth, the hottest event this summer - only
    on MSN
    http://liveearth.msn.com?source=msntaglineliveearthhm

    ---------------------------------------------------------------------
    To unsubscribe, e-mail:
    dev-unsubscribe@sip-communicator.dev.java.net
    <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
    For additional commands, e-mail:
    dev-help@sip-communicator.dev.java.net
    <mailto:dev-help@sip-communicator.dev.java.net>

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To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
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