Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
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@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. On Wed, Aug 20, 2014 at 9:01 AM, <dodul@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----- From: freeswitch-users-request@lists.freeswitch.org Sender: freeswitch-users-bounces@lists.freeswitch.org Date: Wed, 20 Aug 2014 10:06:25 To: <freeswitch-users@lists.freeswitch.org> Reply-To: freeswitch-users@lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 128 Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." _________________________________________________________________________ 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 --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 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@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 _________________________________________________________________________ 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 -- Brian West brian@freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ 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 http://www.freeswitch.org
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.
'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@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@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.
_________________________________________________________________________ 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
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
participants (3)
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Brian West
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Kamrul Khan
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Michael Jerris