[asterisk-users] No compatible codecs, not accepting this offer!

[Gareth Blades]

On 15/01/14 09:39, Francesco Namuri wrote:
> Hello James,
> thanks for your answer, I supposed this too, but my provider answered me
> that as
> m=audio 43718 RTP/AVP 8 18 3 101
>                                               ^  ^  ^---- GSM proposal
>                                               ^  ^------- G729 proposal
>                                               ^---------- aLaw proposal
>
> And that
> a=rtpmap:18 G729/8000  proposed as media conversion
> a=rtpmap:3 GSM/8000/1  because the call is made by a mobile

I would agree with what your service provider has said. If you look at 
the RFC http://tools.ietf.org/html/rfc4566#section-5.14 the '8 18 3 101' 
parameters are a list of media formats. The first is the one which 
should be used but (preferred choice) but the other may be used. Numbers 
in the range 96-127 are dynamic payload types and these must have a 
corresponding 'a=' line specifying the payload type and the codec options.
Lower numbers have static payload assignments and according to that RFC 
dont have to have corresponding 'a=' lines. A list of types can be found 
at http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml

However in all SIP traces I have seen there has always been a 'a=' line 
for every payload type offered. The static payload type numbers are used 
but there is still the 'a=' line.