[Freeswitch-users] Codecs in SDP

[Michael Collins]

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
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
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> >>
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> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
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> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
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> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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