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#581 I can't hear the other party. But they can hear me.

Duplicate
nobody
None
Medium
Defect
2012-05-13
2011-01-10
Anonymous
No

Originally created by: krishnagopal.thogiti

What steps will reproduce the problem?
1.I need to make a call to my sip account.
2.Receive the call. It fully blank call.
3.I tried changing stun and ICE all the trials. It is still not working

It just works for the first time i setup the account, not later.

What is the expected output? What do you see instead?
I should be able to hear the other party. But I am not hearing.

What version of the product are you using? On what operating system?
Android 2.2 and Mytouch 4g.
csipsimple v 0.00-16 [r410]

Please provide any additional information below.

Related

Commit: [r410]
Tickets: #669

Discussion

  • Anonymous

    Anonymous - 2011-01-14

    Originally posted by: r3gis...@gmail.com

    Can you try latest dev version and let me know how it goes?

     
  • Anonymous

    Anonymous - 2011-01-16

    Originally posted by: krishnagopal.thogiti

    This is also not working. Everything works for the first time. But from 2nd or 3rd call onwards, it stops working.

    I am using T-mobile's HSDPA networkig

     
  • Anonymous

    Anonymous - 2011-07-07

    Originally posted by: foxatla...@gmail.com

    Same problem - No audio upon receiving or making a call through GV service, but the other party can hear me.

    CSipSimple ver 0.02-03 [r944];
    Thrive 2.2.1;
    SIP: sipsorcery.com; the SIP number is the forwarding phone number in GV.
    GV is the SIP provider in my sipsorcery account.

    I use the following dial plan in sipsorcery:

    http://google-voice-sipsorcery-dialplans.googlecode.com/svn/trunk/SIP%20Sorcery%20Dial%20Plans/DialPlans/SimpleDialPlan

    The above issue happened frequently during my 2.5 hours' test. About half of the 80 calls got the problem.

    NOTE: 1. There are 2 registrar bindings registered in my sipsorcery account: one user agent is csipsimple [r944]/thunderg-8, and the corresponding contact URI and romote socket use port 27592.
          2. The other, an ATA with the same SIP provider (sipsorcery.com), has been working for a month without any issue.

    HOWEVER, if someone calls me by my sipsorcery SIP number directly, the call quality is very good.

     

    Related

    Commit: [r944]

  • Anonymous

    Anonymous - 2012-05-13

    Originally posted by: r3gis...@gmail.com

    (No comment was entered for this change.)

    Mergedinto: 669
    Status: Duplicate

     

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