[Freeswitch-users] Codecs in SDP

Sat Jun 7 01:00:44 MSD 2014

Yeah, thank you guys, i didn't know that those lines 'a=rtpmap' are
optional. Everything is working as it should now ;-)

2014-06-06 23:46 GMT+03:00 Michael Collins <msc at freeswitch.org>:

> Well, it seems my mail thread management has failed me. I totally missed
> KK's post here. At least we gave him the same info. :P
> -MC
>
>
> On Fri, Jun 6, 2014 at 11:15 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
>
>> The codecs were included from the start (except GSM):
>>
>> m=audio 17262 RTP/AVP 8 0 101 13
>>
>> 8 = PCMA
>> 0 = PCMU
>> 101 = telephone-event (dynamic)
>> 13 = CNG
>>
>> See this:
>>
>> https://wiki.freeswitch.org/wiki/Variable_verbose_sdp
>>
>> You might not be seeing GSM because it's not included on the profile
>> or the module (mod_spandsp, I believe) isn't loaded.
>>
>>
>> On Fri, Jun 6, 2014 at 9:51 AM, Ričardas Stoma <ricardas.stoma at gmail.com>
>> wrote:
>> > It is still the same without single quote marks around PCMA,PCMU,GSM...
>> >
>> > But i think everything is OK with my SDP, i just made a call to my
>> asterisk
>> > server and i see that asterisk receives correct codecs:
>> >
>> > Found audio description format telephone-event for ID 101
>> > Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
>> > audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined -
>> > (gsm|ulaw|alaw)
>> >
>> > So i guess i had to learn more about SDP and codecs before posting my
>> issue,
>> > sorry guys :-)
>> >
>> >
>> > 2014-06-06 15:47 GMT+03:00 Anonio Silva <asilva at wirelessmundi.com>:
>> >
>> >> Can you try:
>> >>
>> >> {absolute_codec_string=PCMA,PCMU,GSM}   ===> without '
>> >>
>> >>
>> >>
>> >> On Fri, 2014-06-06 at 11:36 +0300, Ričardas Stoma wrote:
>> >>
>> >> I have created the following extension:
>> >>
>> >>
>> >> <extension name="my_test">
>> >>   <condition>
>> >>     <action application="bridge"
>> >> data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/
>> 1001 at 192.168.1.78"/>
>> >>     <action application="hangup"/>
>> >>   </condition>
>> >> </extension>
>> >>
>> >>
>> >> then i call to this extension, freeswitch dials to 1001 at 192.168.1.78
>> and
>> >> when i check the content of SDP, included in a SIP packet,  i see
>> something
>> >> like this:
>> >>
>> >>
>> >> v=0
>> >> o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65
>> >> s=FreeSWITCH
>> >> c=IN IP4 192.168.1.65
>> >> t=0 0
>> >> m=audio 17262 RTP/AVP 8 0 101 13
>> >> a=rtpmap:101 telephone-event/8000
>> >> a=fmtp:101 0-16
>> >> a=ptime:20
>> >>
>> >>
>> >> why my codecs from 'absolute_codec_string' are not included in SDP? I
>> >> would expect to see these lines included:
>> >>
>> >> a=rtpmap:8 PCMA/8000
>> >> a=rtpmap:0 PCMU/8000
>> >> a=rtpmap:3 GSM/8000
>> >>
>> >>
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>> >>
>> >> 
>> >> 
>> >>
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>> >>
>> >>
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>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
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>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
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>> >
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>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
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>>
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>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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