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Links: ·ALL ·Review Your VoIP Provider ·VoIP Providers ·VoIP FAQ ·Porting Rules ·What Codec?
AuthorAll Replies

im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

reply to helloagain

Re: [Other] HKBN 2b PAP2 config

(Sorry if I'm blanket bombing all the topics related to HKBN 2b service...this is a common problem for everyone and I can't remember who's tracking which thread...)

I figured out how to get incoming calls working with HKBN 2b.

You need to change the RTP size interval from 0.030 seconds to 0.020 seconds. This config can be found under the admin/advanced settings at "SIP > RTP Parameters > RTP Packet Size" = 0.020. After changing the value, power-cycle the PAP2.

Now, the detailed explanation. This might help others that have problems with immediately dropping connections.

It appears that the Nortel VoIP gateway used by HKBN is not happy when the RTP frame size specified by the Sipura/PAP2 Analog Telephone Adapter (ATA) is not the same as the one it offered.

When an incoming call is received by HKBN, it sends a SIP INVITE to the ATA. Within the INVITE is a Session Description Protocol/Packet (SDP) that describes the characteristics of the media (in this case the audio channel). One of the attributes inside the SDP from HKBN is:
  a=ptime:20
which means "I accept RTP audio frames sized at 20 milliseconds".

The Sipura/PAP2 responds to the SIP INVITE with its own SDP describing how large a frame it accepts. In the case of most Sipura/PAP2 configurations, it's 30 milliseconds. The ATA sends an SDP with the following attribute:
  a=ptime:30

HKBN's Nortel gateway doesn't like this asymmetric frame sizes and aborts the call.

So, setting the ATA's RTP frame size appears to appease the gateway and allows the incoming call's audio channel to passthru to your ATA.

Now that we understand the behavior, we can extrapolate on this. If you are using a VoIP Service Provider that seems to drop the call immediately upon answer by the callee, then check the frame size being offered. If they don't match, then adjust your side and try again.

See ya...

d.c.

NYCLau

join:2005-11-07
New Hyde Park, NY

im_chandave,

I have changed RTP size interval to 0.020 seconds, and I have no problem taking incoming calls now. Thanks for your great advice.


hkvalet
Specialist

join:2004-10-18
Las Vegas, NV

Good job chanave, you helped a lot of other people to solve their problem!!!
--
**Students are the CUSTOMERS of teachers** »RateMyProfessors.com


hkvalet
Specialist

join:2004-10-18
Las Vegas, NV

reply to im_chandave
chandave, are you able to make conference calls and receive call-waiting calls?

I asked you this question before but I don't know if you found the solution or not.

Thanks!!
--
**Students are the CUSTOMERS of teachers** »RateMyProfessors.com


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

Have not been able to get this to work on my SPA or PAP2. The ATAs send the "hold" indication to HKBN. HKBN response back with OK. But, then the ATA sends a "BYE" to terminate the call.

I'll have to record what my asterisk box does and compare the SIP messages.

See ya...

d.c.


dikyee

join:2005-10-31
Vancouver, BC

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


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

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.


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

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.


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

1 edit

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.


dikyee

join:2005-10-31
Vancouver, BC

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


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

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)


Also, an ethereal/tcpdump of the conversation from the softphone side would be useful.

See ya...

d.c.

dikyee

join:2005-10-31
Vancouver, BC

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


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

said by dikyee:

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
»www.voip-info.org/wiki/view/Sipu···slogging

said by dikyee:

Also, how can I get ethereal/tcpdump of the conversation
which you mentioned? Thanks
»www.ethereal.com/
»www.tcpdump.org/

Fastest way to get a TCPdump of the softphone session:
    •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.


See ya...

d.c.

dikyee

join:2005-10-31
Vancouver, BC

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


dikyee

join:2005-10-31
Vancouver, BC

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


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1

1 edit

(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.


dikyee

join:2005-10-31
Vancouver, BC

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


im_chandave

join:2005-07-28
Cleveland, OH
kudos:1
Reviews:
·ViaTalk

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).

Without all that stuff, it's not worth sending the dumps and the logs to me. I won't be able to trace the conversation properly.

Sorry.

See ya...

d.c.

fortissimo

join:2003-10-17
Richmond, BC
Reviews:
·TELUS
·Rynga

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!!


dikyee

join:2005-10-31
Vancouver, BC

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


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