VOIP delay

I initiated a conversation about same problem earlier with little confusion but now I have clear view about the situation.
TP---E1(R2analog)----3660----ethernet--1751 with FXS----TP
I picked 1751 attached telephone handset at 15:42:39.999 (40th sec) pressed digits 15:42:45.739 (45th sec)and got the ringback tone at 15:43:00.(next minuit).This took 15 seconds.
In the debug voip ccapi inoutoutput we can see nothing during 15:42:48.115 & 15:43:00.835 (12 sec) Can anyone help me to reduce this period ?
(Lots of debug lines are ommited because of the limitations of the post)
Apr 22 15:42:46.603: ssaDigit,1. callinfo.called , digit 11, callinfo.calling 6030, xrulecallingtag 0, xrulecalledtag 0
Apr 22 15:42:46.603: ssaDigit, 7. callinfo.calling 6030, sct->digit 11, result 1
Apr 22 15:42:47.139: cc_api_call_digit_begin (dstVdbPtr=0x0, dstCallId=0xFFFFFFFF, srcCallId=0x17,
digit=7, digit_begin_flags=0x1, rtp_timestamp=0xDE0760A4
rtp_expiration=0x0, dest_mask=0x1)
Apr 22 15:42:47.143: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(23), disp(0)
Apr 22 15:42:47.143: cid(23)st(SSA_CS_MAPPING)ev(SSA_EV_DIGIT_BEGIN)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
Apr 22 15:42:47.143: ssaIgnore cid(23), st(SSA_CS_MAPPING),oldst(0), ev(10)
Apr 22 15:42:47.359: cc_api_call_digit_end (dstVdbPtr=0x0, dstCallId=0xFFFFFFFF, srcCallId=0x17,
digit=7,duration=255,xruleCallingTag=0,xruleCalledTag=0, dest_mask=0x1), digit_tone_mode=0
Apr 22 15:42:47.363: sess_appl: ev(9=CC_EV_CALL_DIGIT_END), cid(23), disp(0)
Apr 22 15:42:47.363: cid(23)st(SSA_CS_MAPPING)ev(SSA_EV_CALL_DIGIT)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
Apr 22 15:42:47.363: ssaDigit
Apr 22 15:42:47.363: ssaDigit, 0. sct->digit 11, sct->digit len 2, usrDigit 7, digit_tone_mode=0
Apr 22 15:42:47.363: ssaDigit,1. callinfo.called , digit 117, callinfo.calling 6030, xrulecallingtag 0, xrulecalledtag 0
Apr 22 15:42:47.363: ssaDigit, 7. callinfo.calling 6030, sct->digit 117, result 1
Apr 22 15:42:47.819: cc_api_call_digit_begin (dstVdbPtr=0x0, dstCallId=0xFFFFFFFF, srcCallId=0x17,
digit=9, digit_begin_flags=0x1, rtp_timestamp=0xDE0760A4
rtp_expiration=0x0, dest_mask=0x1)
Apr 22 15:42:47.823: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(23), disp(0)
Apr 22 15:42:47.823: cid(23)st(SSA_CS_MAPPING)ev(SSA_EV_DIGIT_BEGIN)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
Apr 22 15:42:47.823: ssaIgnore cid(23), st(SSA_CS_MAPPING),oldst(0), ev(10)
Apr 22 15:42:48.079: cc_api_call_digit_end (dstVdbPtr=0x0, dstCallId=0xFFFFFFFF, srcCallId=0x17,
digit=9,duration=295,xruleCallingTag=0,xruleCalledTag=0, dest_mask=0x1), digit_tone_mode=0
Apr 22 15:42:48.083: sess_appl: ev(9=CC_EV_CALL_DIGIT_END), cid(23), disp(0)
Apr 22 15:42:48.083: cid(23)st(SSA_CS_MAPPING)ev(SSA_EV_CALL_DIGIT)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
Apr 22 15:42:48.083: ssaDigit
Apr 22 15:42:48.083: ssaDigit, 0. sct->digit 117, sct->digit len 3, usrDigit 9, digit_tone_mode=0
Apr 22 15:42:48.083: ssaDigit,1. callinfo.called , digit 1179, callinfo.calling 6030, xrulecallingtag 0, xrulecalledtag 0
Apr 22 15:42:48.083: ssaDigit, 7. callinfo.calling 6030, sct->digit 1179, result 0
Apr 22 15:42:48.083: ccCallReportDigits (callID=0x17, enable=0x0)
Apr 22 15:42:48.083: cc_api_call_report_digits_done (vdbPtr=0x8227F3A8, callID=0x17, disp=0)
Apr 22 15:42:48.083: ssaSetupPeer cid(23) peer list: tag(3) called number (1179)
Apr 22 15:42:48.087: ssaSetupPeer cid(23), destPat(1179), matched(1), prefix(), peer(822447E8), peer->encapType (2)
Apr 22 15:42:48.087: ccCallProceeding (callID=0x17, prog_ind=0x0)
Apr 22 15:42:48.087: ccCallSetupRequest (Inbound call = 0x17, outbound peer =3, dest=, params=0x8227AAC0 mode=0, *callID=0x8227AEF0, prog_ind = 3) callingIE_present 0
Apr 22 15:42:48.087: ccCallSetupRequest numbering_type 0x81
Apr 22 15:42:48.087: ccCallSetupRequest encapType 2 clid_restrict_disable 1 null_orig_clg 1 clid_transparent 0 callingNumber 6030
Apr 22 15:42:48.087: dest pattern 1..., called 1179, digit_strip 0
Apr 22 15:42:48.087: callingNumber=6030, calledNumber=1179, redirectNumber= display_info= calling_oct3a=0
Apr 22 15:42:48.087: accountNumber=, finalDestFlag=0,
guid=e241.9267.740f.11d7.8030.d508.fd9c.795d
Apr 22 15:42:48.087: peer_tag=3
Apr 22 15:42:48.087: ccIFCallSetupRequestPrivate: (vdbPtr=0x821884F4, dest=, callParams={called=1179,called_oct3=0x81, calling=6030,calling_oct3=0x0, calling_xlated=false, subscriber_type_str=RegularLine, fdest=0, voice_peer_tag=3},mode=0x0) vdbPtr type = 1
Apr 22 15:42:48.087: ccIFCallSetupRequestPrivate: (vdbPtr=0x821884F4, dest=, callParams={called=1179, called_oct3 0x81, calling=6030,calling_oct3 0x0, calling_xlated=false, fdest=0, voice_peer_tag=3}, mode=0x0, xltrc=-5)
Apr 22 15:42:48.091: cc_insert_call_entry: not incoming entry
Apr 22 15:42:48.091: cc_insert_call_entry: entry's incoming FALSE. is_incoming is FALSE
Apr 22 15:42:48.091: ccSaveDialpeerTag (callID=0x17, dialpeer_tag=0x3)
Apr 22 15:42:48.091: ccCallSetContext (callID=0x18, context=0x82457BDC)
Apr 22 15:42:48.091: sess_appl: ev(52=CC_EV_CALL_REPORT_DIGITS_DONE), cid(23), disp(0)
Apr 22 15:42:48.091: cid(23)st(SSA_CS_CALL_SETTING)ev(SSA_EV_CALL_REPORT_DIGITS_DONE)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
Apr 22 15:42:48.091: -cid2(24)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_MAPPING)
Apr 22 15:42:48.091: ssaReportDigitsDone cid(23) peer list: (empty)
Apr 22 15:42:48.091: ssaReportDigitsDone callid=23 Reporting disabled.
Apr 22 15:42:48.095: cc_api_supported_data data_mode=0x10002
Apr 22 15:42:48.095: ccTDUtilGetInstanceCount: For tagID[1] of callID[24]
Apr 22 15:42:48.099: ccTDPvtProfileTableObjectAccessManager: No profileTable set for callID[24]
Apr 22 15:42:48.099: ccTDUtilGetInstanceCount: For tagID[2] of callID[24]
Apr 22 15:42:48.099: ccTDPvtProfileTableObjectAccessManager: No profileTable set for callID[24]
Apr 22 15:42:48.103: cc_incr_if_call_volume: remote IP is 192.168.10.210
Apr 22 15:42:48.103: cc_incr_if_call_volume: hwidb is FastEthernet0/0
Apr 22 15:42:48.103: cc_incr_if_call_volume: create entry in list: 1
Apr 22 15:42:48.111: cc_api_call_proceeding(vdbPtr=0x821884F4, callID=0x18,
prog_ind=0x0, rawmsgPtr=0x0)
Apr 22 15:42:48.111: sess_appl: ev(21=CC_EV_CALL_PROCEEDING), cid(24), disp(0)
Apr 22 15:42:48.115: cid(24)st(SSA_CS_CALL_SETTING)ev(SSA_EV_CALL_PROCEEDING)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(0)fDest(0)
Apr 22 15:42:48.115: -cid2(23)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_CALL_SETTING)
Apr 22 15:42:48.115: ssaCallProc
Apr 22 15:42:48.115: ccGetDialpeerTag (callID=0x17)
Apr 22 15:42:48.115: ssaIgnore cid(24), st(SSA_CS_CALL_SETTING),oldst(1), ev(21)
Apr 22 15:43:00.835: cc_api_call_alert(vdbPtr=0x821884F4, callID=0x18, prog_ind=0x8, sig_ind=0x1)
Apr 22 15:43:00.835: sess_appl: ev(7=CC_EV_CALL_ALERT), cid(24), disp(0)
Apr 22 15:43:00.835: cid(24)st(SSA_CS_CALL_SETTING)ev(SSA_EV_CALL_ALERT)
oldst(SSA_CS_CALL_SETTING)cfid(-1)csize(0)in(0)fDest(0)
Apr 22 15:43:00.835: -cid2(23)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_CALL_SETTING)
Apr 22 15:43:00.835: ssaAlert
Apr 22 15:43:00.835: ccGetDialpeerTag (callID=0x17)
Apr 22 15:43:00.835: ccCallAlert (callID=0x17, prog_ind=0x8, sig_ind=0x1)
Apr 22 15:43:00.835: ccConferenceCreate (confID=0x8227B364, callID1=0x17, callID2=0x18, tag=0x0)
Apr 22 15:43:00.835: cc_api_bridge_done (confID=0x5, srcIF=0x821884F4, srcCallID=0x18, dstCallID=0x17, disposition=0, tag=0x0)
Apr 22 15:43:00.839: cc_api_bridge_done (confID=0x5, srcIF=0x8227F3A8, srcCallID=0x17, dstCallID=0x18, disposition=0, tag=0x0)
Apr 22 15:43:00.839: cc_api_caps_ind (dstVdbPtr=0x821884F4, dstCallId=0x18, srcCallId=0x17,
caps={codec=0xFBFF, fax_rate=0xBF, vad=0x3, modem=0x2
codec_bytes=0, signal_type=3})
Apr 22 15:43:00.839: cc_api_caps_ind (Playout: mode 0, initial 60,min 40, max 300)
Apr 22 15:43:00.839: cc_api_caps_ind (dstVdbPtr=0x8227F3A8, dstCallId=0x17, srcCallId=0x18,
caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0
codec_bytes=20, signal_type=2})
Apr 22 15:43:00.839: cc_api_caps_ind (Playout: mode 0, initial 60,min 40, max 300)
Apr 22 15:43:00.839: cc_api_caps_ack (dstVdbPtr=0x8227F3A8, dstCallId=0x17, srcCallId=0x18,
caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0
codec_bytes=20, signal_type=2, seq_num_start=4683})
Apr 22 15:43:00.839: cc_api_caps_ack (dstVdbPtr=0x821884F4, dstCallId=0x18, srcCallId=0x17,
caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0
codec_bytes=20, signal_type=2, seq_num_start=4683})
Apr 22 15:43:00.843: cc_api_voice_mode_event , callID=0x17
Apr 22 15:43:00.843: Call Pointer =82457668
Apr 22 15:43:00.843: sess_appl: ev(29=CC_EV_CONF_CREATE_DONE), cid(23), disp(0)
Apr 22 15:43:00.843: cid(23)st(SSA_CS_CONFERENCING_ALERT)ev(SSA_EV_CONF_CREATE_DONE)
oldst(SSA_CS_CALL_SETTING)cfid(5)csize(0)in(1)fDest(0)
Apr 22 15:43:00.843: -cid2(24)st2(SSA_CS_CONFERENCING_ALERT)oldst2(SSA_CS_CALL_SETTING)
Apr 22 15:43:00.843: ssaConfCreateDoneAlert
Apr 22 15:43:00.843: sess_appl: ev(50=CC_EV_VOICE_MODE_DONE), cid(23), disp(0)
Apr 22 15:43:00.847: cid(23)st(SSA_CS_CONFERENCED_ALERT)ev(SSA_EV_VOICE_MODE_DONE)
oldst(SSA_CS_CONFERENCING_ALERT)cfid(5)csize(0)in(1)fDest(0)
Apr 22 15:43:00.847: -cid2(24)st2(SSA_CS_CONFERENCED_ALERT)oldst2(SSA_CS_CALL_SETTING)
Apr 22 15:43:00.847: ssaIgnore cid(23), st(SSA_CS_CONFERENCED_ALERT),oldst(4), ev(50)
Apr 22 15:43:00.847: cc_process_notify_bridge_done (event=0x8226B474)
Apr 22 15:43:05.519: cc_api_call_disconnected(vdbPtr=0x8227F3A8, callID=0x17, cause=0x10)
Apr 22 15:43:05.519: sess_appl: ev(11=CC_EV_CALL_DISCONNECTED), cid(23), disp(0)
Apr 22 15:43:05.519: cid(23)st(SSA_CS_CONFERENCED_ALERT)ev(SSA_EV_CALL_DISCONNECTED)
oldst(SSA_CS_CONFERENCED_ALERT)cfid(5)csize(0)in(1)fDest(0)
Apr 22 15:43:05.519: -cid2(24)st2(SSA_CS_CONFERENCED_ALERT)oldst2(SSA_CS_CALL_SETTING)

You are correct ; it is from 1751.
I was calling from the TP attached to the 1751.
3662 seize a channel from E1 to the PBX after this 12 seconds so we can assume the delay is in the 3662 proccesing part.
This does not happen when 1751 replaced by 1750.
(I can post only 10000 words at a time so I edited the configuration.)
3662 config
Busyout is not supported on this voice-port.
Busyout is not supported on this voice-port.
Busyout is not supported on this voice-port.
Busyout is not supported on this voice-port.
version 12.2
voice-card 4
voice call carrier capacity active
voice service pots
voice service voip
voice class codec 1
codec preference 1 g729r8
mta receive maximum-recipients 0
dial-peer voice 2 pots
destination-pattern ....
direct-inward-dial
port 4/0:1
forward-digits all
call rsvp-sync
voice-port 4/0:1
voice-port 4/1:1
mgcp profile default
dial-peer cor custom
dial-peer voice 6030 voip
destination-pattern 03.
session target ipv4:192.168.10.235
tech-prefix 6
alias exec d sh caller
end
1751 configuration
Building configuration...
Current configuration : 1060 bytes
version 12.2
service config
service timestamps debug datetime msec
service timestamps log uptime
no service password-encryption
hostname Router
enable secret xxxxxx
memory-size iomem 15
ip subnet-zero
voice call send-alert
voice rtp send-recv
voice class codec 1
codec preference 1 g729r8
interface FastEthernet0/0
ip address 192.168.10.235 255.255.255.0
speed auto
interface Serial0/0
no ip address
shutdown
interface Serial0/1
no ip address
shutdown
router rip
network 192.168.10.0
ip classless
no ip http server
ip pim bidir-enable
call rsvp-sync
voice-port 2/0
voice-port 2/1
dial-peer cor custom
dial-peer voice 1 pots
destination-pattern 6030
port 2/0
dial-peer voice 3 voip
destination-pattern 1...
voice-class codec 1
session target ipv4:192.168.10.210
dial-peer voice 4 voip
destination-pattern 6...
voice-class codec 1
session target ipv4:192.168.10.210
line con 0
line aux 0
line vty 0
password xxxx
login
line vty 1 4
login
end

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http://www.cisco.com/warp/public/788/voice-qos/voip-ov-fr-qos.html#15
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http://www.cisco.com/warp/public/105/priorityqueuedrops.html
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http://www.cisco.com/warp/public/788/voip/delay-details.html
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http://www.cisco.com/warp/public/788/voice-qos/tos-cos.html
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Hi,
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The system is winxp, sb live , 3 gb ram, celeron 3ghz
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Loops in sound appear randomly and bad sound quality (i give it up javasound)
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Edited by: maxupixu on Mar 26, 2010 8:50 AM

>
> Yes...i see some "?"
I see you making a bunch of declarative statements based on your observations on a system you've not given much technical detail about, and then asking questions that border on being rhetorical.
And I can see a request for a possible solution... You asked for a solution to a problem you're having with an application you didn't remotely talk about how you'd implemented ANYTHING. So how you expect anyone in the world to offer you a solution to your vague babble about 5 to 7 second delays in an application they know nothing about is beyond my comprehension.
Do you think its always required a question to get someone to share some information? I actually walk down the street and crazy people offer to hand me their manifesto, religous people shout out my damnation and/or give me copies of their handouts and religeous texts, I go to a football game and people hand me pamphlets for their political candidate, and I'm often handled fliers with information / meeting times for student organizations on campus. And dude, RSS feeds, man...the RSS feeds just push out information all day long, man. It's amazing! All the information being given to me without asking a single question of anyone / anything...
But yeah, if you want help with something on a coding forum, you'd be better of asking a damned question instead of sharing your observations about what your unposted, undescribed code appears to be doing and expecting someone to share information about where 5 seconds may have came from...
Regardless, I'm going to suggest that a normal program travels significantly closer to the speed of light than your implementation does, and as such, you're experiencing a problem with time dilation. I'd recommend you google around for the special theory of relatively, and make appropriate corrections to your inertial frame of reference. With the amount of information you've presented, that's as good a theory as I can offer.
And I'm officially done with this thread.

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PHONE 3333312---ROUTERA ----DLCI 100----ROUTERB ---PHONE 2111123
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dial-peer voice 1 voip
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frame bc 640
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frame fair-queue 64 256 0
ROUTER B
dial-peer voice 3 voip
destination-pettern 3T
session target ipv4:10.10.10.1
int ser 0/0
encap frame-relay
frame-relay traffic-shaping
no ip address
interface serial0/0.1 point-to-point
ip add 10.10.10.2 255.255.255.252
frame-relay interface-dlci 100
class voice
map-class frame-relay voice
frame cir 64000
frame bc 640
frame mincir 64000
frame-relay ip rtp priority 16384 16383 48
frame fragment 80
frame fair-queue 64 256 0
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12.1(5)T and above should work well.

Help Needed for Resetting VoIP SPA 2100 Codes

Yesterday afternoon, I lost power to my home for a few minutes; until then my phone worked perfectly (for the week that I had it initially installed). I have the Earthlink TrueVoice program with a ZyXEL P 600 series DSL modem and direct Ethernet cable (NO router) to my ATA adapter (Linksys VoIP model SPA 2100). My computer is an HP Pavilion 8755C with Windows Millenium Edition. I have a standard (wire) phone connection. After the power was restored to my home, I tried to use my ATA adapter (Linksys VoIP model SPA 2100) to make a phone call, but there was a five-second delay between when I spoke and when someone on the other end of the line heard me. I contacted Earthlink through online chat and was told to do the following: Dial **** 877778 # 1 to reset the adapter. I did this and nothing positive happened (though I did hear a recorded English woman’s voice repeat the numbers I dialed); I did, however, lose the dial tone and was unable to connect to the Internet with my DSL. So, to conduct more online chat, I re-routed the Ethernet cable to bypass the ATA adapter; I was successful at restoring my Internet, but, obviously, I had NO phone service. More than 8 hours of online chats to Earthlink technical service resulted in failure each time. I swapped ends of my cables numerous times – no success. I removed the power cords to the ATA adapter and to the DSL Modem MULTPILE times for for up to 5 minutes – no success. The lights on the back of the ATA lit properly, but since the first time I tried the **** 877778 # 1 code, the Status light blinks and the light indicating “Phone 1” stays dark (yes, there is astandard phoneline connected to that port). I tried multiple codes to reset the ATA (re-coding instructions are separated by semi-colons): **** 877778 # 1 ; **** 877778 # 1 # ; **** 732668 # ; *** *73738 # 1 . I unplugged the power cord and Ethernet cable (to the modem) on the ATA adapter for about 1 or 2 minutes after trying to reset the code before reattaching everything (power cord always re-attached last). I tried each of these suggestions MULTIPLE times without success. One Earthlink tech wanted more specific information – the MAC address (which I was told is a 12 digit hexa decimal number that starts with zero). I told him what it was. He asked me to try a “ping” test; I was told to click on Start -> Run, type in cmd and hit Enter. A new window would appear and I should type in ping 192.168.0.1 and hit Enter. I never got to the “new window” because after I typed in “cmd” and hit Enter, I received the message “Windows cannot find cmd.” I was told to dial the code * * * * 110 # and to hear my IP address; I heard nothing. I noticed that the last time I tried to dial **** 877778 # 1, I was interrupted before I typed the final “1” because the recorded voice told me “invalid value.” Around 1 a.m. last night I gave up in disgust and went to sleep, disconnecting everything. I have now reconnected the DSL modem and made the direct connection to my computer (the ATA adapter is still disconnected from power and all connecting cables). I am not an idiot and I have some electronics experience (I have wired several home theater systems), but I have little computer experience (and none with VoIP) and right now I am totally frustrated. The tech support at Earthlink is virtually worthless. Please help (and please keep it in simple terms so that an ignorant individual like myself might learn – I need a nice step-by-step method (including times to wait between restarts, what cables/cords to disconnect and the order to disconnect/reconnect them (if that matters) to reset this device) Thanks from a neophyte who is not afraid to be honest and show his ignorance in this area of knowledge.

hi,
i have a SPA-2100 it's a Sipura, and it now asks for a password. I was putting all my settings in for voip provider and now can not get in my admin? i did try the 73738 and that did not work. I was able to put my providers sip and stun all in. is it i'm locked out from my voip provider? If so why would that be, because this is not their device, it's mine? but how do I get back in my admin or can the voip provider I put in, help?
thanks
Message Edited by jokers32463 on 12-22-2008 06:21 PM

Not working properly - - Delay in Video/No audio

Ok so I have a JVC GRDV500 camcorder. I tried to hook the camera up via firewire to chat on iChat with it. However, when I tried to conferences with someone, the video was very slow. It was almost delayed by about 5 seconds. The audio was very terrible as well, and it kept cutting in and out. There is a built in microphone on the camera that should work.
Anyone know a reason as to why the performance is so bad? I would really like to video chat!

Hi,
Umm Bermuda.
Ok Start with.
If anyone is on 10.5.x then go to System Preferences > Quicktime > Streaming at set the drop down to 1.5Mbps and restart iChat (so it "Sees" the new speed)
On Snow Leopard and Leopard computers go to iChat > Preferences > Video Section and set the Bandwidth to 500kbps as a starting point.
If this in itself does nor sort the issue the slowest Buddy (Internet Connection) should drop their's to 200kbps - This is on the iChat Limit in a 3 or 4 way chat as the Specs say 384k
It is worth while having the Connection Doctor (Video Menu) open for this.
Also try changing who is Host.
Ideally it should be the one with the fastest Internet connection.
I have a vague feeling about Bermuda or one of the other small islands where the Local telephone company had become the default ISP - and in the day of iChat 3 where effectively blocking iChat by blocking the SIP Ports (Vonage and other VoIP adapters) that "lost" them Long Distance Telephone connection monies.
The thread about that was some time ago.
It maybe that they cannot support Video chats (Or lots of Off island Video Chats and On-line gaming) due to the ISP's own data rate to the "Main land".
Does it become apparent when you join the chat ?
Or does it not matter who hosts and who joins last ?
8:03 PM Sunday; April 4, 2010
Please, if posting Logs, do not post any Log info after the line "Binary Images for iChat"

How can I use Voip to communicat with my voice in Labview

i want to be able to call my computer using Voip. I know this service
cost money. I dont just want to call the computer, i want to call
labview and all labview to have access to my voice, so i can do speech
commands on it.

Hello,
Thank you for contacting National Instruments. I would like to apologize for the delay in responding to your support request. I understand that you would like to use Voip capabilities to access your computer remotely and then communicate with LabVIEW using speech recognition.
I know of a LabVIEW example file that describes the voice recognition capabilities of LabVIEW. Also, there is a link to example programs within this database entry. To access the entry, go to the National Instruments support portal: http://niweb2.natinst.com/ae/portal.htm and search for �voice recognition LabView.� Click the first entry in the list: �Voice Recognition in LabVIEW - Example - Development Library - National Instruments.� You will be able to find helpful information here.
I hope this helps! Please let me know if I can help you further. Have a great day!
Kind Regards,
Joe Des Rosier
National Instruments

Setting up a VoIP extension on a local network.

With the help of the experts on this board I have successfully set up a VoIP phone extension on our private network. The questions & answers can be viewed at. http://forums.linksys.com/linksys/board/message?board.id=VoIP_Adapters&thread.id=3197 . For the benefit of anyone attempting a similar project, here is the completed setup.
This installation is in a small motel in Te Anau, on New Zealand’s South island. The manager lives off site, and needs to be able to receive calls at night, and also transfer incoming calls to guest’s extensions through the hotels PBX. This necessitates a direct link to the PBX, rather than simply diverting the phone. One solution would have been to lease a circuit from the local Telco, but in NZ, this is very expensive, so another solution was sought. Fortunately there was an established wireless data link between the hotel and the managers residence, so VoIP seemed the obvious choice.
The equipment used is a Linksys SPA3102 connected to an extension on the PBX, and a Linksys PAP2 at the remote end. The setup would work equally well if connected to a phone line, rather than the PBX.
I’ll start the setup with the SPA3102.
Connect the POTS line to the LINE port, and your switch/router to the INTERNET port. In my setup the Ethernet port is not used. Plug a standard phone into the Phone port. This is useful for testing and setting up. It’s not needed afterwards, unless you want a local phone.
Open your web browser, and type the adaptor IP into the address bar. Go to Admin, and Advanced Settings.
ROUTER SETUP
WAN Setup Tab:
Connection type: Static IP.
Static IP Settings: The Network address on your local network (192.168.x.x)
Subnet mask 255.255.255.0
LAN Setup Tab:
LAN IP address: This is automatically selected to be on a different sub net from the WAN. Unless it conflicts with another address on your system you shouldn’t change it.
Enable DHCP: No
(Save these settings.)
VOICE SETUP
System Tab: No Changes
SIP Tab: No Changes
Provisioning Tab: No Changes
Regional Tab: Mostly this sets the dial tones etc to match your local service. Unless you need them to be the same this shouldn’t need any changes
The Hook Flash Timer Min & Max: should be set to the local values. The Defaults (.1 and .9) are OK for North America. Australia and New Zealand use .07 & .13. If you have trouble sending a hook flash, check these values against the local settings.
DTMF playback level should be greater than zero. (I used 3)
(Save these settings)
Line 1 Tab:
I don’t use Line 1 except for testing. During setup the line should be enabled. After the system is running OK, it can be disabled
Line enabled yes
SIP port 5060
Proxies are not used in this setup.
Register: No
Make call without reg: yes
Answer call without reg: yes
User ID: 10? (you can use any number)
Line 1 Tabupplementary services.
Change Call waiting, 3 way Conf, and 3 way call, to no. (These interfere with sending a hook flash)
Hook Flash Tx method: AVT
(save these settings)
PSTN Line Tab
Line enable: yes
SIP Port 5061 (default)
Proxy: proxies are not used.
Register: no
Make call w/o reg yes
Answer call w/o reg yes
Display name: anything you like (VoIP gateway?)
User ID: leave blank
User password: leave blank
Use auth ID: no
Dial Plan 1: (<:*>S0). Switches to the outside line when * received.
Dial Plan 2: (<:[email protected]:5060>S0). 11 is the user ID on the PAP2
VoIP to PSTN enable: yes
VoIP caller default DP: 1
One stage Dialing: no
VoIP users & Passwords.
User 1 ID: 11. User1 DP: 1
User 2 ID: 21 User 2 DP: 1
User 3 ID 22 User 3 DP: 1
(These are the line numbers of additional PAP2’s on our system)
PSTN to VOIP Gateway enable: yes
PSTN Caller ID none
PSTN Caller Default DP: 2
Detect PSTN long silence yes
Detect VoIP long silence yes
Detect Disconnect tone yes
VoIP answer delay 0
PSTN Answer delay 0
PSTN to VoIP gain (Set these to adjust
VoIP to PSTN gain the speech volume)
Line in Use voltage: This should be set midway between the On Hook and Off Hook voltages, which you get from the Info screen. Most public phones are 47v on hook, and 7v off hook, so the setting should be 27v. My PBX is 27v on hook, and 7v off hook, so my setting is 17v. To read this, go to the Info screen and check the Line Voltage, then go Off hook (make a call), click the reload button on your browser, and check the line voltage again.
(save these settings)
This completes the setting up of the SPA3102.
Now for the setup of the PAP2.
Open your web browser, and type the PAP2 IP into the address bar. Go to Admin, and Advanced Settings.
System tab:
DHCP no
Static IP 192.168.x.x (same sub-net as your network. Different adaptor number)
Net Mask 255.255.255.0
(save these settings)
SIP Tab: no changes.
Provisioning Tab: no Changes
Regional Tab.
Hook Flash Min & Max: change to your local settings if required.
(save these settings)
Line 1 & Line 2 Tabs.
Whether you use Line 2 depends on whether you want to have 2 phones on the PAP2. All calls from the PSTN line of the SPA3102 will go to Line 1 of the PAP2 as per Dial Plan 2 on the SPA
Line enable yes
SIP port 5060 (line 1) & 5061 (line 2)
Proxy Proxies are not used.
Register no
Make call w/o reg yes
Answer call w/o reg yes
Display name: anything you like
User ID 11 (line 1) & 12 (line 2)
(These are used to identify each line on the system)
Call waiting: no
3 way conf: no
3 way call: no
DTMF Tx method: AVT
Dial Plan: This is the dial plan I use on line 1.
(<:192.168.4.10:5061>S3|21S0<:@192.168.4.9:5060>|22S0<:@192.168.4.9:5061>)
You will have to modify it for use on other lines, or other adaptors, and the IP addresses must match your system IP addresses. Here is an explanation.
192.168.4.10:50613 All my adaptors are on subnet 4. 10 is the number of the SPA3102, and 5061 is the SIP port mapped to the PSTN line. If the handset is lifted, and no numbers are dialed the call will be transferred to the PSTN line after 3 seconds, and you will hear the outside dial tone. If within 3 seconds you dial either 21, or 22, the phone on either line 1, SIP port 5060, or Line 2, SIP port 5061, on adaptor 9 will ring. (If you only have one PAP2 then you will only need the first section of this dial plan.)
Enable IP Dialing: yes
(save these settings).
User 1 and User 2 tabs: no changes
That just about does it. All incoming calls from outside are received by the PBX, and after hours are sent to the extension connected to the SPA3102, which rings the phone on the remote PAP2 in the manager’s house. If the call is for a guest we can press the recall button (hook flash), dial the guest’s extension number, and transfer the call when they answer. As an added bonus we have a second PAP2 elsewhere on the network, and we can call between the 3 adaptors. All 3 adaptors have access to an outside line, though the PBX. I’m fairly sure it would also work through a VPN, which would mean we could take a VoIP phone anywhere in the world, and still be virtually ‘On site". I don’t know if that is a good thing or not.

Hi HW,
The PBX is a Panasonic TA308. There is no special interface to the PBX, the line port on the SPA3102 is simply plugged into an extension, like another phone. Anyone calling that extension will have the call routed through the SPA & PAP2 to the remote phone.
The whole setup is totally seamless, & transparent to the user. As we are on a local network there is virtually no latency. There is a slight tendancy to echo, but the echo suppression mostly takes care of that.
THis has been a good exercise, and once I got my head around what I was trying to do, with your help, it was pretty easy. I think the hook flash timing would be the thing which gives most users a problem, as it seems to vary widely around the world. I was surprised at the difference between the US and NZ (.1 & .9 to .07 & .13). There didn't seem to be any other critical differences.
Now I am the local expert on VoIP "In the Kingdom of the blind, the one-eyed man is King."

Delay in video and Audio

I use iChat to video chat with two friends in the UK. One is a o2 user and the other is a talktalk user. Wen chatting we experience an up to 7 seconds delay in voice and Video and audio. When we change to Skype there is no delay. Can you explain and solve this please. You do a great job Ralph. Thank you.

Hi,
Umm Bermuda.
Ok Start with.
If anyone is on 10.5.x then go to System Preferences > Quicktime > Streaming at set the drop down to 1.5Mbps and restart iChat (so it "Sees" the new speed)
On Snow Leopard and Leopard computers go to iChat > Preferences > Video Section and set the Bandwidth to 500kbps as a starting point.
If this in itself does nor sort the issue the slowest Buddy (Internet Connection) should drop their's to 200kbps - This is on the iChat Limit in a 3 or 4 way chat as the Specs say 384k
It is worth while having the Connection Doctor (Video Menu) open for this.
Also try changing who is Host.
Ideally it should be the one with the fastest Internet connection.
I have a vague feeling about Bermuda or one of the other small islands where the Local telephone company had become the default ISP - and in the day of iChat 3 where effectively blocking iChat by blocking the SIP Ports (Vonage and other VoIP adapters) that "lost" them Long Distance Telephone connection monies.
The thread about that was some time ago.
It maybe that they cannot support Video chats (Or lots of Off island Video Chats and On-line gaming) due to the ISP's own data rate to the "Main land".
Does it become apparent when you join the chat ?
Or does it not matter who hosts and who joins last ?
8:03 PM Sunday; April 4, 2010
Please, if posting Logs, do not post any Log info after the line "Binary Images for iChat"

QOS - VOIP traffic: payload and signalling

Two questions For VOIP traffic,
Q.1. should the payload and signalling be assigned the same COS ?? to avoid losing Signalling traffic.
Q.2. flash upgrades to the phone sets are tftp. Should this traffic be assigned COS=0 or the same cos as the signalling traffic ? Phone flash gets corrupted if some of these packets are lost ?
thanks for your help,

Hello,
best practice puts signaling into another class than voip "payload" as you named it.
The reason is that the qos requirements are different. Signaling needs guaranteed bandwidth, voip needs low delay and guaranteed bandwidth.
TFTP can be placed into signaling class or in a separate class.
An example config could look like this:
ip cef
class-map match-any VoIP
match ip dscp ef cs5
match protocol rtp audio
class-map match-any VoIPsignal
match ip dscp cs3
class-map match-all TFTP
match protocol tftp
policy-map VoIPprio
class VoIP
priority percent 10
class VoIPsignal
bandwidth remaining percent 5
class TFTP
bandwidth remaining percent 5
class class-default
fair-queue
random-detect
interface Serial0
ip address ...
service-policy output VoIPprio
You would have to adjust the bandwith ratios for your needs and the interface naming to your environment.
The policy will give 10% of the link to voip traffic with lowest possible delay, 5% of the rest for voip signaling, 5% to TFTP and the rest for the other data present.
Hope this helps

VoIP Environment with Many Subscribers

Hello,
I have the following environment setup:
- 3 Aironet AP (AIR-LAP 1242AG)
- 1 WLAN controller (WLC 2106)
- VoIP device, e.g. IP phone, beltpack
- Security policies [WPA + WPA2][Auth(PSK)]
Please advise:
1. What is the maximum number of subscribers can be supported per AP in one BSS?
2. Why is it limited to that number of subscribers?
3. What is the best choice of IEEE standards for VoIP (IEEE 802.11a, 802.11b or 802.11g)?
4. If we have 1000 subscribers in one big studio in one IP subnet (one WLC), please advise how many Cisco APs must be allocated?
5. What are the things to consider to avoid interference, voice delay and the best way to assign or divide the channel allocation for each AP?
Thank you.

Hi,
There is no magic answer to your question without a proper site survey and without knowing your clients specifications, but here are some common values (provided that a site survey was conducted properly) if you were using Cisco 7921:
1. Up to 27 active subscribers in ideal conditions, but as real world is not ideal conditions, aiming at 14 clients on 802.11g, or 7/8 clients on 802.11b/g, 20 clients on 802.11a. This is again IF you did a good site survey and respected the recommended practices. This document will help you:
http://www.cisco.com/en/US/docs/solutions/Enterprise/Mobility/vowlan/41dg/vowlan41dg-book.html
2. The reasons for this limitation is the wireless space. Each user needs to send a certain number of voice packets per second, so you don't have space for more than a certain number of users without facing the risk of too many packets collisions, retries, and their corresponding delay and jitter issues.
3. 802.11a is usually quieter (less sources of interference) than 802.11/b/g, and is usually preferred for voice deployment... but it also depends on you clients! Are they 802.11a ? b? g? This will dictate what you can do.
4. 1000 users in one subnet? This looks very high to me... this is one broadcast domain, I would split them into smaller subnets... but on the wireless side, if you do the math (and if as usual you do a clean site survey, which will also need to take into account the expected density of users in each area), you would deploy somewhere around 50 APs if you deploy in 802.11a (but this also depends very much on how "big" your studio is, and how many APs you need to cover the physical area).
5. To avoid interferences, many things can be done, the very first one being a site survey. Then, there are usually more channels and less interferences in 802.11a. Then read the document referenced above, there are many things you need to do and understand to do a proper VoWLAN deployment...
Hope it helps
Jerome

How to develop VoIP client using SIP in J2ME?

Hi Everybody,
I want to develop a VoIP client in J2ME that connects to asterisk server of debian and can call to the registered user of asterisk server and can have a telephonic talk session easily.
Do anybody have idea regarding the development of the client or having tutorial that teaches the development of VoIP in J2ME or in any other way.?
PLZ help me to provide the solution.
Thanks in anticipation.
with regards,
KHAKHAR SAGAR

Hi
I am interested about developing VoIP application (using SIP) in J2ME platform. But I am stuck with the problem of MMAPI. Without using MMAPI J2ME has no access to mobile media devices, such as speaker or microphone, and without creating a player MMAPI can't play media data, such as sound or video. But its not possible to record voice and play voice data simultaneously using player in J2ME. So it seems almost impossible to implement VoIP application maintaining all its constraints and requirements, specially in case of delay and jitter.
I am looking for some solution, which will provide the ability to overcome this problem. I come out with two possible solutions, but not sure about their out come. If we can develop a native media application, we can have access to it by using KNI (K Native Interface). In that way we can take some risk to develop VoIP application for J2ME. My another solution is, we can handle the player using MMAPI to record and play voice data in mill second level, so that we can have a real time feeling, though I am not sure if its possible by using RTSP.
If any one have solution of this problem, please help us.
Reagards
Asif Mohammed Adnan

Delay with PAP2T connected with Intercom

Hi guys,
maybe you have some hints for me.
At the office entrance we've got a doorbell/intercom.
If you ring - the intercom connects to the voip telephone system.
We connected the RJ11 of the intercom to the PAP2T.
So far, the system's running but the problem is that the delay between pressing the bell and getting the call is about 10 to 15 seconds.
I already tested that the call goes at once to the PAP2T but then it takes time til the voip-server gets the signal and connects to the specified phone. So i can definitively blame the PAP2T for the lag ;-)
Do you have any ideas what settings i should take a look on.
I'm thankful for any help.

Hi RRamirez,
if possible, can you terminate the sequence sent to PAP2T with "#" ?
I mean, wanna to dial 320, let the sequence 320# been sent to PAP2T and watch results.
Hope this helps.

Unacceptable delay on Gizmo Call Out

Why do I get unacceptable delay (latency) using Gizmo Call Out with a 811g router? When using SkypeOut from my laptop through the same router, the latency is fine.
I am using a Nokia E51 and I have the same results whether I call international or local numbers.
I have tried both D-Link and LinkSys routers with the same results.

Do you use Sat Internet? Do you have latency when using Gizmo Call Out with the same laptop?
Generally, latency issues are caused by the internet path between you and the destination. You can test your own connection for voip quality using an online testing service like testyourvoip.com. This test displays your TO and FROM latency MOS and Round-Trip Latency.
If the results from this are good (Round-Trip below 300ms), do a ping test from your laptop to proxy01.sipphone.com and compare the Round-Trip time to the results of the online test.
If all is good from your laptop, let me know and we continue checking for other issues that may be causing latency on your Nokia E51. Thanks.
-GizmoAce
Message Edited by gizmoace on 22-May-2008 01:01 AM

VOIP delay

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