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@32000h@20i,speex@16000h@20i,speex@8000h@20i,iLBC@30i,G7221@32000h,G7221@16000h,opus,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@32000h@20i,speex@16000h@20i,speex@8000h@20i,iLBC@30i,G7221@32000h,G7221@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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