[Freeswitch-users] Codec list truncation?

[Keith Laaks]

Hi,

Are you perhaps hitting the MTU limit?

-- 
Keith Laaks 
 
--------------



On 2014/03/21, 6:06 AM, "Pete Ashdown" <pashdown at xmission.com> wrote:

>Is there some sort of limit in my SIP rtpmap for codecs?  I've got this
>list of codecs:
>
>  <X-PRE-PROCESS cmd="set"
>data="global_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@20i
>,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/>
>  <X-PRE-PROCESS cmd="set"
>data="outbound_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@2
>0i,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/>
>
>I'm seeing this output from tcpdump:
>
>    v=0
>    o=FreeSWITCH 1395348540 1395348541 IN IP4 10.10.10.1
>    s=FreeSWITCH
>    c=IN IP4 10.10.10.1
>    t=0 0
>    m=audio 25672 RTP/AVP 0 98 99 100 102 103 104 105 8 3 101 13
>    a=rtpmap:98 SPEEX/32000
>    a=rtpmap:99 SPEEX/16000
>    a=rtpmap:100 SPEEX/8000
>    a=rtpmap:102 iLBC/8000
>    a=fmtp:102 mode=30
>    a=rtpmap:103 G7221/32000
>    a=fmtp:103 bitrate=48000
>    a=rtpmap:104 G7221/16000
>    a=fmtp:104 bitr[|sip]
>
>
>Note the last line with bitr[|sip] where bitrate should be.  This causes
>phones that would otherwise answer the call with the available codecs
>above to ignore and not ring at all.
>
>_________________________________________________________________________
>