I have a freeswitch server located behind 1:1 nat that I have been using to test things with. Today I upgraded to the latest trunk and began having troubles with my private ip being exposed in the sdp instead of the value set with ext-rtp-ip in the sip_profile. Needless to say all of my audio is broken now. :) I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this. -Dale
You need to set the local-network-acl on the profile too please refer to the default configs. /b On Jul 29, 2009, at 11:30 AM, Dale wrote:
I have a freeswitch server located behind 1:1 nat that I have been using to test things with. Today I upgraded to the latest trunk and began having troubles with my private ip being exposed in the sdp instead of the value set with ext-rtp-ip in the sip_profile. Needless to say all of my audio is broken now. :)
I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this.
-Dale
Hmm, just tried that by adding the <param name="local-network-acl" value="localnet.auto"/> to the profile. I also verified that the acl was working properly. I just tried defaulting back to the stock sample configs and setting the ext-rtp-ip to my public ip and I still see the private in the sdp when I dial the echo test. -Dale On Jul 29, 2009, at 12:39 PM, Brian West wrote:
You need to set the local-network-acl on the profile too please refer to the default configs.
/b
On Jul 29, 2009, at 11:30 AM, Dale wrote:
I have a freeswitch server located behind 1:1 nat that I have been using to test things with. Today I upgraded to the latest trunk and began having troubles with my private ip being exposed in the sdp instead of the value set with ext-rtp-ip in the sip_profile. Needless to say all of my audio is broken now. :)
I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this.
-Dale
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Are you dialing the echo test from a public IP or a private IP? btw you need to set the ext-sip-ip too. /b On Jul 29, 2009, at 12:25 PM, Dale wrote:
Hmm, just tried that by adding the
<param name="local-network-acl" value="localnet.auto"/>
to the profile. I also verified that the acl was working properly.
I just tried defaulting back to the stock sample configs and setting the ext-rtp-ip to my public ip and I still see the private in the sdp when I dial the echo test.
-Dale
What do you mean? Is the client behind NAT? if so what client? and where is the IP in the SDP? the one from the client or the one from FreeSWITCH and what is the network layout of the scenario because that is critical to understanding what the heck is going on here. /b On Jul 29, 2009, at 12:52 PM, Dale wrote:
Are you dialing the echo test from a public IP or a private IP? btw
Its a client natted behind a public ip. I don't have a pure public IP to test from at the moment.
you need to set the ext-sip-ip too
Yep, got that set too.
The box is currently setup with the default config(and password) if you want to try registering to it and placing a call. FS (172.16.50.10) -> NAT (Public IP 207.136.252.50) -> INTERNET -> NAT (Public IP 64.9.127.104) -> Zoiper(Soft Phone) (10.10.10.147) Freeswitch offers SDP v=0 o=FreeSWITCH 1248869138 1248869139 IN IP4 172.16.50.10 s=FreeSWITCH c=IN IP4 172.16.50.10 t=0 0 m=audio 20054 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Zoiper offers SDP v=0 o=Z 0 0 IN IP4 64.9.127.104 s=Z c=IN IP4 64.9.127.104 t=0 0 m=audio 16000 RTP/AVP 110 98 0 8 3 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 On Jul 29, 2009, at 1:55 PM, Brian West wrote:
What do you mean? Is the client behind NAT? if so what client? and where is the IP in the SDP? the one from the client or the one from FreeSWITCH and what is the network layout of the scenario because that is critical to understanding what the heck is going on here.
/b
On Jul 29, 2009, at 12:52 PM, Dale wrote:
Are you dialing the echo test from a public IP or a private IP? btw
Its a client natted behind a public ip. I don't have a pure public IP to test from at the moment.
you need to set the ext-sip-ip too
Yep, got that set too.
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Are you on SVN trunk? I can tell you 100% that this has to be a config error. Are you talking over the right profile? /b On Jul 29, 2009, at 1:10 PM, Dale wrote:
The box is currently setup with the default config(and password) if you want to try registering to it and placing a call.
FS (172.16.50.10) -> NAT (Public IP 207.136.252.50) -> INTERNET -> NAT (Public IP 64.9.127.104) -> Zoiper(Soft Phone) (10.10.10.147)
Freeswitch offers SDP
v=0 o=FreeSWITCH 1248869138 1248869139 IN IP4 172.16.50.10 s=FreeSWITCH c=IN IP4 172.16.50.10 t=0 0 m=audio 20054 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
Zoiper offers SDP
v=0 o=Z 0 0 IN IP4 64.9.127.104 s=Z c=IN IP4 64.9.127.104 t=0 0 m=audio 16000 RTP/AVP 110 98 0 8 3 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000
I'm running FreeSWITCH Version 1.0.trunk (14412) Okay, I'll keep hunting around. -Dale On Jul 29, 2009, at 2:21 PM, Brian West wrote:
Are you on SVN trunk? I can tell you 100% that this has to be a config error. Are you talking over the right profile?
/b
On Jul 29, 2009, at 1:10 PM, Dale wrote:
The box is currently setup with the default config(and password) if you want to try registering to it and placing a call.
FS (172.16.50.10) -> NAT (Public IP 207.136.252.50) -> INTERNET -> NAT (Public IP 64.9.127.104) -> Zoiper(Soft Phone) (10.10.10.147)
Freeswitch offers SDP
v=0 o=FreeSWITCH 1248869138 1248869139 IN IP4 172.16.50.10 s=FreeSWITCH c=IN IP4 172.16.50.10 t=0 0 m=audio 20054 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
Zoiper offers SDP
v=0 o=Z 0 0 IN IP4 64.9.127.104 s=Z c=IN IP4 64.9.127.104 t=0 0 m=audio 16000 RTP/AVP 110 98 0 8 3 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000
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Still stumped on trying to write a streamlined pocketsphinx speech recognition. I went through SpeechTools.jm and I think I pulled out the necessary pieces, but the speech event is not being fired. I get dtmf events if I push a key, but not speech. I must be missing a piece, but I have stared at SpeechTools.jm until I can't see straight. Here is my code: function onInput(s, type, inputEvent, _this) { console_log("debug", "EVENT:" + type + "\n"); } session.answer(); var blankobj = new Object(); var rv; var hit; var dup; var grammar_name="pizza_yesno"; var collected_index=0; var req=1; var obj_path="result"; var halt=true; var min_score=20; var confirm_score=10; console_log('INFO', "Setting grammar\n"); session.execute("detect_speech", "nogrammar " + grammar_name); session.execute("detect_speech", "pocketsphinx " + grammar_name); session.execute("detect_speech", "resume"); console_log('INFO', "entering while loop\n"); session.streamFile( "audiofile.wav", onInput, "blankobj") ; while(session.ready() && collected_index < req) { var x; if (!rv) { rv = session.collectInput(onInput, "blankobj", 500); if (rv && !rv[0]) { rv = false; } } if (!rv) { session.execute("detect_speech", "resume"); rv = session.collectInput(onInput, "blankobj", 5000); } hit = false; if (rv) { var items = rv; rv = undefined; for (y = 0; y < items.length; y++) { console_log("debug", "---- "+items[y]+"\n"); } } if (!rv) { rv = session.collectInput(onInput, "blankobj", 1000); } } //////////////////////////////////////////// //////////////////////////////////////////// And here is some of the output from the console: 2009-07-29 16:10:09 [INFO] stest-examp.js:1 console_log() Setting grammar EXECUTE sofia/internal/+*@199.173.94.88:5060 detect_speech(nogrammar pizza_yesno) EXECUTE sofia/internal/+*@199.173.94.88:5060 detect_speech(pocketsphinx pizza_yesno) EXECUTE sofia/internal/+*@199.173.94.88:5060 detect_speech(resume) 2009-07-29 16:10:09 [INFO] stest-examp.js:1 console_log() entering while loop 2009-07-29 16:10:09 [DEBUG] switch_ivr_play_say.c:1084 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-07-29 16:10:09 [DEBUG] switch_core_io.c:649 switch_core_session_write_frame() sofia/internal/+*@199.173.94.88:5060 receive message [TRANSCODING_NECESSARY] 2009-07-29 16:10:09 [DEBUG] sofia.c:2979 sofia_handle_sip_i_state() Channel sofia/internal/+*@199.173.94.88:5060 entering state [ready][200] 2009-07-29 16:10:12 [DEBUG] switch_ivr_play_say.c:1379 switch_ivr_play_file() done playing file EXECUTE sofia/internal/+*@199.173.94.88:5060 detect_speech(resume) 2009-07-29 16:10:16 [DEBUG] switch_rtp.c:1876 switch_rtp_dequeue_dtmf() RTP RECV DTMF 5:2440 2009-07-29 16:10:16 [DEBUG] stest-examp.js:45 console_log() EVENT:dtmf EXECUTE sofia/internal/+*@199.173.94.88:5060 detect_speech(resume) EXECUTE sofia/internal/+*@199.173.94.88:5060 detect_speech(resume) 2009-07-29 16:10:27 [DEBUG] switch_rtp.c:1876 switch_rtp_dequeue_dtmf() RTP RECV DTMF 5:2320 2009-07-29 16:10:27 [DEBUG] stest-examp.js:45 console_log() EVENT:dtmf Greg
SpeechTools.jm is rather simple if you look at ps_pizza.js that uses it.. you'll get a better feel for how that works. Have you looked at the pizza demo javascript? /b On Jul 29, 2009, at 3:28 PM, Greg Thoen wrote:
Still stumped on trying to write a streamlined pocketsphinx speech recognition. I went through SpeechTools.jm and I think I pulled out the necessary pieces, but the speech event is not being fired. I get dtmf events if I push a key, but not speech. I must be missing a piece, but I have stared at SpeechTools.jm until I can't see straight. Here is my code:
Hi. Yes, I have had that running, no problem. I agree that SpeechTools as a class is easy to use, but I am trying to write a smaller version that I can integrate more with what I am doing; so I am trying to pull out pieces of it. In the example I gave, I am concerned that I can't even get my onInput to see the event. I feel I must be missing some step in my code. -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 On Jul 29, 2009, at 4:34 PM, Brian West wrote:
SpeechTools.jm is rather simple if you look at ps_pizza.js that uses it.. you'll get a better feel for how that works. Have you looked at the pizza demo javascript?
/b
On Jul 29, 2009, at 3:28 PM, Greg Thoen wrote:
Still stumped on trying to write a streamlined pocketsphinx speech recognition. I went through SpeechTools.jm and I think I pulled out the necessary pieces, but the speech event is not being fired. I get dtmf events if I push a key, but not speech. I must be missing a piece, but I have stared at SpeechTools.jm until I can't see straight. Here is my code:
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http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts /b On Jul 29, 2009, at 4:05 PM, Greg Thoen wrote:
Hi. Yes, I have had that running, no problem.
I agree that SpeechTools as a class is easy to use, but I am trying to write a smaller version that I can integrate more with what I am doing; so I am trying to pull out pieces of it. In the example I gave, I am concerned that I can't even get my onInput to see the event. I feel I must be missing some step in my code. -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260
I have the same issue with my freeswitch server running on the Amazon EC2 instance, I set up the ext-rtp-ip address with the public routable IP address but the private IP address showing up in the SDP. When I checked my home router's internet sessions the rtp sessions were built using Amazon EC2 private IP address. Chris On Wed, Jul 29, 2009 at 12:30 PM, Dale <fdhege@gmail.com> wrote:
I have a freeswitch server located behind 1:1 nat that I have been using to test things with. Today I upgraded to the latest trunk and began having troubles with my private ip being exposed in the sdp instead of the value set with ext-rtp-ip in the sip_profile. Needless to say all of my audio is broken now. :)
I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this.
-Dale
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participants (4)
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Brian West
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Chris Chen
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Dale
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Greg Thoen