[Freeswitch-users] Codec list truncation?

[Peter Steinbach]

Hello Pete,

could you check the packet size while wiresharking the traffic? When a
larger number of codecs is enabled, the SIP packet size can exceed the
MTU size (~1500 bytes), so the single packet received is not complete.
FS should send then 2 SIP UDP packets, but most SIP devices cannot
assemble them. When possible, you may switch to TCP then when submitting
theses large SIP packets. Freeswitch can handle this.

/Peter

On 03/21/14 05:06, Pete Ashdown 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@20i,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.
>
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-- 
With kind regards
Peter Steinbach 

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