[Freeswitch-users] Sofia late-negotiation on re-INVITE (codec-modification)

[Paul Cupis]

On 27/11/11 18:10, Phil Quesinberry wrote:
> A friend of mine dialed into one of my conference rooms with only the GSM
> codec enabled on his mobile SIP client (Linphone).  Despite the fact that I
> offered GSM as one of the codecs, FS only appeared to offer G711u and the
> call failed.  When he set G711u as an option on his end, he was able to
> connect.  Late negotiation is enabled.

The SDP from the mobile SIP client is bogus, it offers codec 0 (PCMU) 
but tries to redefine it as GSM.

FreeSWITCH accepts the offer of PCMU and negotiates it - but then the 
callers client sends/expects invalid media:

    m=audio 9003 RTP/AVP 0 0 3 101
    a=rtpmap:0 GSM/22050
    a=rtpmap:0 GSM/11025
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-11

http://www.iana.org/assignments/rtp-parameters

0 PCMU/8000
3 GSM/8000

It should be using 96-126 for GSM at other bitrates...

Regards,