[Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
Brian West
brian at freeswitch.org
Thu Aug 21 00:31:18 MSD 2014
'show codecs' isn't the list of codecs you've allowed on your sofia
profile, its just a listing of all registered codecs. I would recommend
you do 'sofia status profile xxx' and look at the CODECS IN/CODECS OUT.
On Wed, Aug 20, 2014 at 11:28 AM, Michael Jerris <mike at jerris.com> wrote:
> 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 at 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 at gmail.com
> To: freeswitch-users at 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.
>
>
>
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--
*Brian West*
brian at freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
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