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

[James Sharp]

On 1/15/2014 5:50 AM, Gareth Blades wrote:
> 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.
>

I missed the RTP/AVP line.  I'll go back to lurking for a bit :)