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



--Forwarded Message Attachment--
From: matt@inveroak.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 20 Aug 2014 11:59:35 +0100
Subject: Re: [Freeswitch-users] Internal IP showing in SDP

Hi Brian,

it seems that this was not the issue.

Some more details:
We have an opensips server using the load_balancer module to route the calls to the freeswitch box. (https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS)
When I route the calls directly to the freeswitch's external IP, the calls connect fine and have audio.
However when I route the calls to the opensips server first, the call will connect but without audio.

The strange thing is Opensips uses Freeswitch's internal address (from the load_balancer table) to forward the INVITE packet, yet when the ACK packet comes in from the provider opensips tries to forward it to the FS external IP address rather than the internal IP.

I understand this is probably more an opensips question, so will understand if I should post this to their mailing list, but any tips would be appreciated.


thanks
Matt

thanks brian, will give it a go and see what happens.

thanks
Matt

On 19/08/2014 17:06, Brian West wrote:

Can you try prefixing the ext-rtp-ip and ext-sip-ip with autonat:0.0.0.0  and make sure local-network-acl is set to rfc1918.auto?

On Tue, Aug 19, 2014 at 9:20 AM, Matt Broad <matt@inveroak.com> wrote:

Hi,

does it make any difference if my internal IP address is being sent in
the SDP connection information?  I have 2 sip providers where 1 I can
bridge calls and receive audio and 1 that I can bridge calls but get no
audio.

Even with the below settings the SDP message body contains c=IN IP4
192.168.1.1.

I have vars set as:
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.ip.goes.here"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=my.ip.goes.here"/>

I have setup the external profile as follows:
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>

running sofia status profile external gives (replace 0.0.0.0 with the
real external IP declared in vars):
=================================================================================================
Name                    external
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  192.168.1.1
Ext-RTP-IP              0.0.0.0
SIP-IP                  192.168.1.1
Ext-SIP-IP              0.0.0.0
URL                     sip:mod_sofia@0.0.0.0:5080
BIND-URL sip:mod_sofia@0.0.0.0:5080;maddr=192.168.1.1;transport=udp,tcp
WS-BIND-URL             sip:mod_sofia@192.168.1.1:5086;transport=ws
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G722,PCMU,PCMA,GSM
CODECS OUT              PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           true
AGGRESSIVENAT           false
CALLS-IN                8
FAILED-CALLS-IN         1
CALLS-OUT               8
FAILED-CALLS-OUT        1
REGISTRATIONS           0




thanks
Matt


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--

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brian@freeswitch.org


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T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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