The freeswitch debug logs should tell you more about what is happening. I would suggest turning on sip trace as well. Mike On Aug 20, 2014, at 11:19 AM, Kamrul Khan <dodul@live.com> wrote:
Hi, Thanks for your reply. In that case, how to know what exactly the issue is. The requested codecs in my INVITE is as follows: a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:actpass a=candidate:0 1 UDP 2128609535 172.16.1.188 56772 typ host a=candidate:2 1 UDP 1692467199 184.69.59.132 56772 typ srflx raddr 172.16.1.188 rport 56772 a=candidate:5 1 UDP 2128543999 192.168.56.1 56773 typ host a=candidate:10 1 UDP 2128478463 192.168.232.1 56774 typ host a=candidate:15 1 UDP 2128412927 192.168.146.1 56775 typ host a=candidate:0 2 UDP 2128609534 172.16.1.188 56776 typ host a=candidate:1 2 UDP 1692467198 184.69.59.132 56776 typ srflx raddr 172.16.1.188 rport 56776 a=candidate:5 2 UDP 2128543998 192.168.56.1 56777 typ host a=candidate:10 2 UDP 2128478462 192.168.232.1 56778 typ host a=candidate:15 2 UDP 2128412926 192.168.146.1 56779 typ host a=rtcp-mux The codec installed in my freeswitch: show codecs type,name,ikey codec,ADPCM (IMA),mod_voipcodecs codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_voipcodecs codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_voipcodecs codec,G.726 16k (AAL2),mod_voipcodecs codec,G.726 24k,mod_voipcodecs codec,G.726 24k (AAL2),mod_voipcodecs codec,G.726 32k,mod_voipcodecs codec,G.726 32k (AAL2),mod_voipcodecs codec,G.726 40k,mod_voipcodecs codec,G.726 40k (AAL2),mod_voipcodecs codec,G.729,mod_g729 codec,GSM,mod_voipcodecs codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_voipcodecs codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
Is is rejecting just because I dont have opus installed in my freeswitch? But, I do have ulaw and alaw. The same configuration works fine with asterisk and I dont have OPUS installed there aswell. Asterisk codec list: ID TYPE NAME DESCRIPTION -------------------------------------------------------------------------------- --- 100001 audio g723 (G.723.1) 100002 audio gsm (GSM) 100003 audio ulaw (G.711 u-law) 100004 audio alaw (G.711 A-law) 100011 audio g726 (G.726 RFC3551) 100006 audio adpcm (ADPCM) 100019 audio slin (16 bit Signed Linear PCM) 100007 audio lpc10 (LPC10) 100008 audio g729 (G.729A) 100009 audio speex (SpeeX) 100016 audio speex16 (SpeeX 16khz) 100010 audio ilbc (iLBC) 100005 audio g726aal2 (G.726 AAL2) 100012 audio g722 (G722) 100021 audio slin16 (16 bit Signed Linear PCM (16kHz)) 300001 image jpeg (JPEG image) 300002 image png (PNG image) 200001 video h261 (H.261 Video) 200002 video h263 (H.263 Video) 200003 video h263p (H.263+ Video) 200004 video h264 (H.264 Video) 200005 video mpeg4 (MPEG4 Video) 400001 text red (T.140 Realtime Text with redundancy) 400002 text t140 (Passthrough T.140 Realtime Text) 100013 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 100014 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) 100017 audio testlaw (G.711 test-law) 100015 audio g719 (ITU G.719) 100028 audio speex32 (SpeeX 32khz) 100020 audio slin12 (16 bit Signed Linear PCM (12kHz)) 100022 audio slin24 (16 bit Signed Linear PCM (24kHz)) 100023 audio slin32 (16 bit Signed Linear PCM (32kHz)) 100024 audio slin44 (16 bit Signed Linear PCM (44kHz)) 100025 audio slin48 (16 bit Signed Linear PCM (48kHz)) 100026 audio slin96 (16 bit Signed Linear PCM (96kHz)) 100027 audio slin192 (16 bit Signed Linear PCM (192kHz)) 100018 audio silk8 (SILK Custom Format 8khz) 100018 audio silk12 (SILK Custom Format 12khz) 100018 audio silk16 (SILK Custom Format 16khz) 100018 audio silk24 (SILK Custom Format 24khz)
--Forwarded Message Attachment-- From: sertys@gmail.com To: freeswitch-users@lists.freeswitch.org Date: Wed, 20 Aug 2014 12:43:11 +0200 Subject: Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
He's implying that something else apart from the SRTP part is giving you the 488. Like an inability to negotiate a proper codec for the call.