On Fri, Jul 31, 2009 at 7:41 PM, Jordan L. Walbesser <
[email protected]> wrote:
> This is definitely a known bug. There's a patch for it, I know I submitted
> one a while back, but I doubt it got included into a build. Someone else
> wrote this up too.
>
> See the following:
> http://osdir.com/ml/telephony.sipp.user/2008-07/msg00148.html
>
> Let me know if you need some more guidance.
>
> ~Jordan
> --
> Jordan L. Walbesser
> Computer Engineering B.S. '07
> Juris Doctor '10
>
> Mobile: (716) 640-0532
> Web: www.buffalo.edu/~jlw28 <http://www.buffalo.edu/%7Ejlw28>
>
>
>
> On Fri, Jul 31, 2009 at 7:20 PM, Bradley, Todd
> <[email protected]>wrote:
>
>> I'm running SIPp on both Windows (using pre-built binaries) and Linux
>> (64 bit CentOS system, built from source). I have a scenario where I'm
>> sending audio and video media from my SIPp scenario to a video
>> conferencing system.
>>
>> The scenario works fine under Windows, but I was wondering why the audio
>> and video aren't played when I run the scenario under Linux. Using
>> Wireshark, I found that SIPp is not sending the audio and video RTP
>> packets to the port numbers specified in the SDP. Instead, it appears
>> to be truncating those numbers.
>>
>> In the 200 OK message from the UAC, the SDP says to use port 49160 for
>> audio media and port 49170 for video media. However, the actual RTP
>> packets are being sent to ports 4916 and 4917, according to Wireshark.
>>
>> Has anyone seen this, and do you have a workaround or patch for it?
>>
>>
>> Thanks,
>> Todd.
>> ----
>> Todd Bradley - Polycom, Inc.
>> DMA 7000 QA Lead
>>
>>
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