[Asterisk-Dev] silencesupp header in SDP

[Atif Rasheed]

I think omitting ast_build_string() call for media attribute annexb=no 
from chan_sip.c has nothing to do with audio. as commenting/removing 
this call will just remove the below three headers from outgoing SDP 
messages. should not effect two-way RTP bridge.

However, I have tested a single call which worked for me. I will test it 
with bunch of calls, but I am sure that it will not create any problems.

Atif

Moises Silva wrote:

>Youre right, chan_sip.c is the right place. But AFAIK Asterisk does
>not support silence suppression. If you remove that asterisk may stop
>sending sound, because it seems it uses  the incoming voice frames
>(including silence frames) as timer to send its own sound frames. 
>Plase correct me if im wrong.
>
>Regards
>
>On 1/17/06, Atif Rasheed <atif at iphonica.com> wrote:
>  
>
>>Sorry, its not silencesupp attribute, its Annex-B indication which was
>>not in 1.0.9 and now its there in 1.2.1 and probably here parsing is
>>failing.
>>
>>Media Attribute ( a ): fmtp:18 annexb=no
>>                 Media Attribute Fieldname: fmtp
>>                 Media Attribute Value: 18 annexb=no
>>
>>
>>Shoud I remove it from chan_sip.c, i.e. I just remove this block
>> if (codec == AST_FORMAT_G729A)
>>                /* Indicate that we don't support VAD (G.729 annex B) */
>>                ast_build_string(a_buf, a_size, "a=fmtp:%d
>>annexb=no\r\n", rtp_code);
>>
>>and Annex-B indication will not be there in outgoing SDP messages ???
>>
>>Thank you
>>
>>Atif
>>
>>Atif Rasheed wrote:
>>
>>    
>>
>>>hello all,
>>>
>>>How can I strip this header from outgoing SDP messages
>>>
>>>Media Attribute (a): silenceSupp:off - - - -
>>>               Media Attribute Fieldname: silenceSupp
>>>               Media Attribute Value: off - - - -
>>>
>>>probably it was not there in 1.0.9, but when I upgraded to 1.2.1 its
>>>creating problems for some GW's .
>>>
>>>Thank you
>>>
>>>Atif
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>
>
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