VOIP etc > Voice Over IP - VOIP > VOIP Tech Chat > [Other] HKBN 2b PAP2 config

reply to hkvalet

Re: [Other] HKBN 2b PAP2 config

Hi chandave!
I give you a 2 thumbs up on the RTP size solution. But it comes with another minor problem,is that callers from any 2b softphone to my PAP2 can't hear my voice after the phone connected.
It seems to us that this is the only problem that left for you, mighty to fix on HKBN.Awaiting for your good news.

Tks

Can you guys at least help me get the Syslog dumps of a problem session? I don't normally use a PAP2 with HKBN. As a result, I have to keep on tearing down my current configuration and re-setup the PAP2 VoIP environment.

Actually, what would be great would be the detailed syslog as well as a tcpdump/ethereal capture of the I/O between the softphone HKBN while attempting to connect to a HKBN subscriber using a PAP2.

See ya...

d.c.

reply to dikyee
Can you guys at least help me get the Syslog dumps of a problem session? I don't normally use a PAP2 with HKBN. As a result, I have to keep on tearing down my current configuration and re-setup the PAP2 VoIP environment.

Actually, what would be great would be the detailed syslog as well as a tcpdump/ethereal capture of the I/O between the softphone HKBN while attempting to connect to a HKBN subscriber using a PAP2.

See ya...

d.c.

reply to dikyee
Does anyone else confirm this is a problem with them? And, can you guys at least help me get the Syslog dumps of a problem session? I don't normally use a PAP2 with HKBN. As a result, I have to keep on tearing down my current configuration and re-setup the PAP2 VoIP environment.

Actually, what would be great would be the detailed syslog as well as a tcpdump/ethereal capture of the I/O between the softphone HKBN while attempting to connect to a HKBN subscriber using a PAP2.

See ya...

d.c.

Hi Chandave!
Tks for your speedy response.Here is the log you requested,I hope this could help .Thanks once again

[0]Off Hook
[0:5060]->203.80.89.139:5060
[0:5060]->203.80.89.139:5060
SIP/2.0 200 OK

t: "HKBN HKBN" ;tag=64d655d489c985di0

f: "HKBN HKBN" ;tag=1b4c-538-b40-f3a7e838

i: 107f3a55999174bd2a745fd9f75fc8c54073f56faf@203.80.89.139

CSeq: 1 INVITE

v: SIP/2.0/UDP 203.80.89.139:5060;branch=z9hG4bK4f63017c77e05e305b3b3c54a6dd975b

Contact: Anonymous

Server: Linksys/PAP2-3.1.7(LSe)

Content-Length: 235

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: x-sipura

Content-Type: application/sdp

v=0

o=- 6807970 6807970 IN IP4 24.87.188.36

s=-

c=IN IP4 24.87.188.36

t=0 0

m=audio 16460 RTP/AVP 8 100 101

a=rtpmap:8 PCMA/8000

a=rtpmap:100 NSE/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

I need the whole conversation:

•SIP INVITE from HKBN
•PAP2 100/183 Trying
•PAP2 200 OK
•PAP2 declaration of the RTP ports
•BYE (from either PAP2 or HKBN)
•100/183 Trying (from either PAP2 or HKBN)
•200 OK (from either PAP2 or HKBN)

I've only obtained such brief log from the software I'm using.Too briefly that I couldn't find datas you requested.
Sorry can you suggest me any better syslog capture software that would help me on this
Also, how can I get ethereal/tcpdump of the conversation
which you mentioned? Thanks

•Get Windows Pcap libraries
•Get Windows TCPDump program
•Install WinPcap
•At a Windows CMD.EXE issue the command line:


windump.exe -D

A list of devices should show up. Remember the number of the device that matches your ethernet interface. In this example let's assume the number was '2'.
•At the Windows CMD.EXE issue the command line:


windump.exe -i 2 -l -s 1500 -w hkbn.dmp (host s21.hkbntel.net or host s22.hkbntel.net) and (port 5060)

The '2' of '-i 2' should be caused to the number of the ethernet interface found in the 'windump -D' step.
•Startup your softphone, call the number associated with the PAP2. Let the PAP2 pickup the call. Run with the dead air call for at least 5 seconds. Hangup the PAP2.
•Terminate the windump by hitting <Ctrl>-C.
•You can see what you captured by:


windump -l -s 1500 -v -r hkbn.dmp

1. Post your session dump (it's the hkbn.dmp file) or IM it to me.
   

Hi Dave!
As those thingie are new to me, I don't want to make mistake, will do it on weekend & let you know. Thanks

rgds

Hi Dave
Regarding the windump.exe what should I issue per your instruction #5? as it shows nothing on the forum page.
Kindly re-send details by e-mail to sunny3322@hotmail.com
Thanks

(aahhhggggg.... MS IE can't render my posting properly...)

windump.exe -i 2 -l -s 1500 -w hkbn.dmp (host s21.hkbntel.net or host s22.hkbntel.net) and (port 5060)

It should be all on one line.

See ya...

d.c.

Tks.have sent you the dmp report via IM,please check .tks

I'm sorry Sunny. The stuff you cut and pasted into the IM you sent me did not have any information I can use to diagnose the problem. I really need the binary capture file generated by the 'windump -w hkbn.dmp ...' command.

Additionally, the traffic captures really need to have all the stuff I was asking you to capture. The stuff needed for the syslog dump from the PAP2 can be found in this posting: http://www.broadbandreports.com/forum/remark,14588534~days=9999~start=40;iframe=1#14942708. The stuff for the tcpdump would be:

1. SIP INVITE from your softphone to HKBN
   
2. HKBN 100/183 Trying
   
3. HKBN 200 OK
   
4. HKBN declaration of the RTP ports
   
5. BYE (from either HKBN or the softphone
   
6. 100/183 Trying (from either HKBN or the softphone)
   
7. 200 OK (from either HKBN or softphone).
   

reply to dikyee
Sorry has been away a little.

My friend and I experienced the problem that dikyee has, which is some calls have no audio, IIRC, it's the calls from (may be to????) a HKBN 2b softphone. Actually, we experienced similar problem with vBuzzer as well.

Haven't got a dump yet, but will try. Sorry didn't help you guys as fast as I wanted.

Recap the other problem: Sipura 3000, Line 1 on HKBN 2b, PSTN Line on another provider. User 1 set w/ Forward when busy and Forward when no answer, both to a cellular number on @gw0 . A friend got his working fine, but with no PSTN Line setting at all. My friend who has a config on PSTN Line, has a wierd problem: When the call was not answered, it'll drop to PSTN and call out the cellular number. It'll ring, and both sides can hear the other side, for about 10 seconds, then one side will hear beeep-beep-beep, and then the call will not have audio on either side. Strangest thing and yet very repeatable (consistent), unless I fiddle with the delay for the Forwarding, which makes it get less "voice connection time", but never better.

I've got 2 Sipura 3000 w/ me now, and I'll do a config swap around to test. Then I'll wipe out PSTN Line config, and mimic my other friend's setting to see if it helps. Will also try to get a dump, now I read how it's done.

Thanks for all the help im_chandave!!

Hi fortissimo!
I was trying very hard to capture the dump datas that Chandave need to investigate the problem.However everytime I end Dave the report, turns out it's incorrectly.I'm kind of giving it up.Could you try it & send all datas to Dave,he said he will find out the solution for us. Tks

reply to im_chandave
Hi Dave,

Happy new year, long time no see, as my hkbn was working for outgoing calls :P, however, it was not working again, I got busy signal when I have any outgoing call right after I punched in the phone number, do you have any suggestion? my firmware for my ata was sipura-2000 3.1.5, here is my log for the busy signal for every outgoing calls:

PLEASE HELP ;)

RSE_DEBUG: reference domain:s21.hkbntel.net
[0:5060]->203.80.89.135:5060
[0:5060]->203.80.89.135:5060
INVITE sip:92448888@s2hkbntel.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.8:5060;branch=z9hG4bK-f516699c
From: ;tag=f8fdd4d54e47c034o0
To:
Call-ID: af3b5629-f10e0ef0@192.168.2.8
CSeq: 101 INVITE
Max-Forwards: 70
Contact:
Expires: 240
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 229
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 58502 58502 IN IP4 192.168.2.8
s=-
c=IN IP4 192.168.2.8
t=0 0
m=audio 19388 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

[0:5060]
f: ;tag=f8fdd4d54e47c034o0
i: af3b5629-f10e0ef0@192.168.2.8
CSeq: 101 INVITE
v: SIP/2.0/UDP 192.168.2.8:5060;received=61.18.11.60;branch=z9hG4bK-f516699c
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0

[0:5060];tag=324660314
f: ;tag=f8fdd4d54e47c034o0
i: af3b5629-f10e0ef0@192.168.2.8
CSeq: 101 INVITE
v: SIP/2.0/UDP 192.168.2.8:5060;received=61.18.11.60;branch=z9hG4bK-f516699c
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0

[0:5060]->203.80.89.135:5060
[0:5060]->203.80.89.135:5060
ACK sip:92448888@s2hkbntel.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.8:5060;branch=z9hG4bK-f516699c
From: ;tag=f8fdd4d54e47c034o0
To: ;tag=324660314
Call-ID: af3b5629-f10e0ef0@192.168.2.8
CSeq: 101 ACK
Max-Forwards: 70
Contact:
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0

[0:0]AUD Rel Call
CC:Failed
[0]On Hook
RSE_DEBUG: unref domain, s21.hkbntel.net
Sess Terminated
RSE_DEBUG: unref domain, s21.hkbntel.net
RSE_DEBUG: last unref for domain s21.hkbntel.net
Sess Terminated
CC:Clean Up
--- OBJ POOL STAT ---
OP:RTPRXB = 96 ( 96 192) OP:RTPREB = 40 ( 40 48)
OP:RTPTXB = 64 ( 64 108) OP:TIMEOU = 111 (120 40)
OP:SIPCOR = 0 ( 1 28) OP:SIPCTS = 32 ( 32 564)
OP:SIPSTS = 32 ( 32 3452) OP:SIPAUS = 6 ( 8 588)
OP:SIPDLG = 10 ( 10 140) OP:SIPSES = 12 ( 12 8184)
OP:SIPREG = 3 ( 4 244) OP:SIPLIN = 0 ( 2 128)
OP:STUNTS = 16 ( 16 68)
RSE_DEBUG: reference domain:s21.hkbntel.net
[0:5060]->203.80.89.135:5060
[0:5060]->203.80.89.135:5060
REGISTER sip:s2hkbntel.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.8:5060;branch=z9hG4bK-4f4b7b74
From: ;tag=65b11c9361af07d7o0
To:
Call-ID: 14269483-323e70a7@192.168.2.8
CSeq: 9 REGISTER
Max-Forwards: 70
Proxy-Authorization: Digest username="35961323",realm="Realm",nonce="MTEzNjA4OTE
xODMwOTE2ZTRiOTk2ZTY4MzBhODJjMDY3NmNmOTA3Njg5MGZk",uri="sip:s2hkbntel.net",algor
ithm=MD5,response="a7d5a2d8c5757bafe692ec7b44cd5aa1",qop=auth,nc=00000004,cnonce
="a1dbedcc"
Contact: ;expires=120
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

•Don't try removing information unless you are sure what you removed is not relevant. You removed a lot of info in the Contact:, RESPONSES, and other things that makes it difficult to understand what is being responded by what.
•I haven't tried using HKBN on an i-Cable connection yet. An associate of mine attempted it but couldn't succeed. Of couse, she only tried once and didn't want to try to troubleshoot it.

•A SIP INVITE from your Sipura without any credentials being sent to HKBN
•What might be a SIP response of 100 Trying from HKBN
•What might be a SIP response of 200 OK from HKBN
•Your Sipura's ACK'ing the 200 response
•A failed establishment of audio channels since you told HKBN to use the IP address of 192.168.2.8 (your internal pre-NAT address) instead of your i-Cable address.
•A SIP REGISTER from your Sipura to HKBN with the proper credentials...but it appears cut-off and no response from HKBN was supplied in the syslog dump.

•Turn on STUN support
•Enable NAT Mapping

• Use 'X's for the digits of your phone number in the syslog dumps
  
• Use 'Y's for the digits of the party you are calling or is calling you
  
• Change the digits in "Digest username=" to 'X's
  
• Leave the IP addresses in the "Via:" [or "v:" when receiving the SIP message from HKBN] lines. It very important for diagnostics. If you must sanitize it, just change the last digits of the "received=" IP address
  
• Change the digits in the "From:" [or "f:" when receiving the SIP message from HKBN] to 'X's or 'Y's depending on if it's a call from you or going to you. Leave everything else in the line the same
  
• Change the digits in the "To:" [or "t:" when receiving the SIP message from HKBN] to 'X's or 'Y's depending on if it's a call from you or going to you. Leave everything else in the line the same
  
• Change the digits in the "Contact:" [or "m:" when receiving the SIP message from HKBN] to 'X's or 'Y's depending on if it's a call from you or going to you. Leave everything else in the line the same
  
• Don't change the "Call-ID:" [or "i:" when receiving the SIP message from HKBN] line at all.