[Freeswitch-users] Freeswitch srtp rejecting SRTP with 488

Wed Aug 20 20:28:26 MSD 2014

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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