[Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
Kamrul Khan
dodul at live.com
Wed Aug 20 19:19:36 MSD 2014
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/48000/2a=ptime:20a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 56772 typ hosta=candidate:2 1 UDP 1692467199 184.69.59.132 56772 typ srflx raddr 172.16.1.188 rport 56772a=candidate:5 1 UDP 2128543999 192.168.56.1 56773 typ hosta=candidate:10 1 UDP 2128478463 192.168.232.1 56774 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 56775 typ hosta=candidate:0 2 UDP 2128609534 172.16.1.188 56776 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 56776 typ srflx raddr 172.16.1.188 rport 56776a=candidate:5 2 UDP 2128543998 192.168.56.1 56777 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 56778 typ hosta=candidate:15 2 UDP 2128412926 192.168.146.1 56779 typ hosta=rtcp-muxThe 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.
On Wed, Aug 20, 2014 at 9:01 AM, <dodul at live.com> wrote:
Hello,
Thanks for your reply. I'm sorry I guess didn't get you.
Are you saying the log I provided in my earlier email is incomplete?
Or I have to check some other log? If that is so can you please clarify which log? (Eg log name)
Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part?
Thanks in advanced :)
Sent from my “contract free” BlackBerry® smartphone on the WIND network.
-----Original Message-----
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Date: Wed, 20 Aug 2014 10:06:25
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From: matt at inveroak.com
To: freeswitch-users at 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
On 19/08/2014 17:25, Matt Broad wrote:
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 at 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 at 0.0.0.0:5080
BIND-URL sip:mod_sofia at 0.0.0.0:5080;maddr=192.168.1.1;transport=udp,tcp
WS-BIND-URL sip:mod_sofia at 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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