[Asterisk-Users] RTP.C codec error 19

[Jeremy McNamara]

Um no.   Turn off Silence suppression (VAD) on your endpoint.


Jeremy McNamara



Lord Stroud wrote:

>Hi Dave,
>
>  The RTP codec 19 error that you are getting indicates that your endpoint is
>most probably activating silence supression, and that you are using a codec
>such as g.729, at least, that is what I get on my platform here.
>
>  You can go into the the rtc.c file, and simply comment out the message.
>Edit the rtp.c file at line 330, as the following:
>
>ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
>
>  simply edit it to be:
>
>//ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
>
>  and simply re-compile.
>
>Nir S
>
>On Tuesday 08 July 2003 04:42 pm, Dave Alan Caruana wrote:
>  
>
>>hi ..
>>when placing a SIP call to a sip host in the states
>>every few seconds I get an RTP codec 19 error.
>>I know this is related to comfort noise, and the
>>call goes through OK ... how can I suppress
>>the error message ?
>>
>>Also, many times I get "Invalid CSeq Number"
>>back from 216.52.153.207 (which is the host
>>i'm calling) and the call drops.. is there a solution
>>for this ?
>>
>>cheers
>>Dave
>>
>>(I mistakenly put an "re" in the title of this email
>> and I think it's been ignored .. reposted)
>>
>>
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>>
>
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