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Most likely the problem with not being able to hear is due to SDP negotiation. wxCommunicator does not use the same IDs for codecs like the other side (it is not mandated) which may confuse other softphones. You can check it with wireshark, set filter to "sip" so that you only see sip communication. Then find INVITE (with SDP offer) and 200 OK (with SDP answer). See some SIP tutorial...
2009-10-11 17:40:08 UTC by yaro_23
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Hello
Please read https://sourceforge.net/projects/wxcommunicator/forums/forum/689556/topic/1960166
I don't know which version of sipXtapi you want to use, but if you want to use trunk from sipXtapi google then you actually have to use Intel IPP 6.0. 5.3 only works with older source code.
In sipXtapi google mailinglist you can find a post from 7.1.2009 23:48 titled "sipxtapi with...
2009-10-11 17:36:15 UTC by yaro_23
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The customized-sipxtapi source code need some head files in ipp_5_3_project_files.zip(https://sourceforge.net/projects/wxcommunicator/files/).
However, how to use the ipp_5_3_project_files.zip file ? I downloaded this zip file and unzipped in one folder. When compiling projects in unzipped folder with visual studio 2008, there are errors like "Cannot open file ***.c". Furthermore...
2009-10-11 15:53:36 UTC by justanywhere
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Hello
When I use wxcommunicator with a bind IP address of 0.0.0.0 I cannot call hard sip phones directly by ip ( ie sip:10.10.10.2 ). If I put the actual ip address in the bind address ( ie 10.10.10.1 ) then it works just fine calling a hard IP phone. The problem is that the person in the hard phone can hear me just fine, but I cannot hear them. I have tried it with 3 different brand ip...
2009-10-01 19:57:37 UTC by jdpfl
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Unfortunately there is no way to manually adjust the jitter buffer in wxCommunicator/sipXtapi. It is adjusted automatically based on observed jitter. But perhaps your problem is not jitter but packet loss? When you run "ping -n 100 www.google.com" do you see packet loss? Packet loss starvation of jitter buffer could cause sound being skipped every 5-10s.
The latest wxCommunicator...
2009-09-22 11:11:00 UTC by yaro_23
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I've tested on Windows XP and Windows 2000 machines with same results.
I've tried audio device config changes but still hear same problem. I've also tried Ekiga and noticed that it also has same problem but not as frequent as wxCommunicator (every couple minutes instead of every 5-10 seconds). Ekiga shows that PCMU codec is used by default and its GUI shows line statistics and I can see tha.
2009-09-22 04:18:46 UTC by vedamaker
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I recommend to try changing input/output audio device in configuration->audio->audio devices. Do you also hear these sound problems if you record your call and then play the wav file? Do you use Windows XP or Vista?.
2009-09-20 08:37:06 UTC by yaro_23
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I am trying v1.0.5 on Windows against freeswitch PBX to check out the sound quality. I noticed that the sound quality is improved when I set the priority of the wxCommunicator to "HIGH". However, there is still periodic sound drop out every 5-10 seconds. In comparison, X-lite has no such problem in same test. Any idea on how to fix?.
2009-09-19 16:30:10 UTC by vedamaker
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yaro_23 added trilexcom to the wxCommunicator project.
2009-08-28 13:45:03 UTC by yaro_23
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Hello yaro,
thank you for your support, this is the sipxtapi log for one (maybe 2) call I hope it can be useful.
"2009-07-28T13:19:54.390000":2:SIPXTAPI:DEBUG:teres1::00000000:sipXtapi:"ENTER FUNC (tid=188) sipxConfigGetVersion\n"
"2009-07-28T13:19:54.390000":3:SIPXTAPI:DEBUG:teres1::00000000:sipXtapi:"EXIT FUNC (tid=188) sipxConfigGetVersion\n"...
2009-08-06 15:39:11 UTC by dan180
