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Regovich, Timothy

2004-Feb-11 09:42 UTC

[Asterisk-Users] Can't connect KPhone to asterisk

Not ACK'ing an invite can be problematic for the statemachine.  Without the
ACK, the Dialog is not in  acorrect state.

As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
is doing the same.  There is no restriction that they must match.  You can
change your offer in the ACK, or with a re-INVITE.

As for the immediate transmission : yeah, it does seem a little strange
doesn't it?  But that is the way that I have seen almost all UAs work.  The
implication is that your offer must be a valid, not a conditional offer :
when you say you accept GSM on port 8000, you better have a listener on 800
ready to go. 

Tim

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Maciek Kaminski
Sent: Wednesday, February 11, 2004 11:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can't connect KPhone to asterisk


Anyone managed to make KPhone work with Asterisk?

For me it looks as if KPhone does not ACK transactions, i.e.:

KPhone --INVITE--> Asterisk
Asterisk --Trying --> KPhone
Asterisk --OK --> KPhone
KPhone doest not acknowlege. Asterisk keeps resending OKs, KPhone 
INVITES. Both timeouts.

By the way: KPhone offers PCMU, GSM, iLBC in INVITE, Asterisk answers 
with PCMU and PCMA with doest not seem to be correct as it should answer 
with subset of codecs offered(as far as I understood SIP RFC). Another 
issue that bothers me is that Asterisk seems to start media transmission 
as soon as it send OK not after it received ACK. Begining of 
conversation may lost this way, isn't it?

Asterisk and KPhone logs below:

----------------------------------------------------------------------------
-----------------
Asterisk log:
----------------------------------------------------------------------------
-----------------
Sip read:
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

11 headers, 9 lines
Using latest request as basis request
Sending to 192.168.0.3 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format GSM
Found description format iLBC
Capabilities: us - 12, them - 1030/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:4186 check_user: Setting NAT on 
RTP to 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:5277 handle_request: Check for 
res for maciejka
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:1128 find_user: Call from user 
'maciejka' is 1 out of 0
Looking for 700 in default
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:3572 build_route: build_route: 
Contact hop: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>
list_route: hop: <sip:maciejka@192.168.0.3;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Length: 0


 to 192.168.0.3:5060
    -- Executing Answer("SIP/maciejka-b4b6", "") in new
stack
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.0.3:5060
    -- Executing Festival("SIP/maciejka-b4b6", "Press 1 to heaven
press
2 to go to hell press 3 to disconnect.") in new stack
  == Parsing '/etc/asterisk/festival.conf': Found
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:318 festival_exec: Text 
passed to festival server : Press 1 to heaven press 2 to go to hell 
press 3 to disconnect.
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:395 festival_exec: Passing 
text to festival...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:414 festival_exec: Passing 
data to channel...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:424 festival_exec: 
Festival WV command


Sip read:
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

11 headers, 9 lines
Ignoring this request
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3364 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.0.3:5060
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.0.3:5060
  == Spawn extension (default, 700, 2) exited non-zero on 
'SIP/maciejka-b4b6'
Feb 11 17:12:37 DEBUG[180236]: chan_sip.c:1207 sip_hangup: 
find_user(maciejka) - decrement inUse counter
set_destination: Parsing <sip:maciejka@192.168.0.3;transport=udp> for 
address/port to send to
set_destination: set destination to 192.168.0.3, port 5060
Reliably Transmitting:
BYE sip:maciejka@192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK77cfdb1c
From: <sip:700@polimorfia>;tag=as3b0a9ff0
To: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Contact: <sip:700@192.168.0.2>
Call-ID: 1239913767@192.168.0.3
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.0.3:5060

----------------------------------------------------------------------------
-----------------
KPhone log:
----------------------------------------------------------------------------
-----------------

SipClient: Sending: 17:18:40.048
--------------------------------
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

SipClient: Sending to '192.168.0.2:5060'
SipClient: Receiving message...

SipClient: Received: 17:18:40.168
---------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Length: 0


SipClient: Receiving message...

SipClient: Received: 17:18:40.182
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

SipTransaction: Retransmit 1 (500)

SipClient: Sending: 17:18:40.551
--------------------------------
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

SipClient: Receiving message...

SipClient: Received: 17:18:40.568
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3364 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

SipClient: Receiving message...




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States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as Banyu)
that may be confidential, proprietary copyrighted and/or legally privileged. It
is intended solely for the use of the individual or entity named on this
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in error, please notify us immediately by reply e-mail and then delete it from
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Maciek Kaminski

2004-Feb-11 10:10 UTC

head link

[Asterisk-Users] Can't connect KPhone to asterisk

Regovich, Timothy wrote:
>Not ACK'ing an invite can be problematic for the statemachine.  Without
the
>ACK, the Dialog is not in  acorrect state.
>
>As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
>is doing the same.  There is no restriction that they must match.  You can
>change your offer in the ACK, or with a re-INVITE.
>  
>Well, they must intersect:
"For streams marked as sendrecv in the answer, the "m=" line MUST
contain at least one codec the answerer is willing to both send and 
receive, from amongst those listed in the offer. The stream MAY indicate 
additional media formats, not listed in the corresponding stream in the 
offer, that the answerer is willing to send or receive (of course, it 
will not be able to send them at this time, since it was not listed in 
the offer).
>As for the immediate transmission : yeah, it does seem a little strange
>doesn't it?  But that is the way that I have seen almost all UAs work. 
The
>implication is that your offer must be a valid, not a conditional offer :
>when you say you accept GSM on port 8000, you better have a listener on 800
>ready to go. 
>  
>Optimistic strategy...

Maciek Kaminski

Regovich, Timothy

2004-Feb-11 11:46 UTC

head link

[Asterisk-Users] Can't connect KPhone to asterisk

Where is that quote from?
Are rtpmaps marked as sendrecv or recvonly?
There is nothing really that says that I couldn't receive mpeg audio, but
only be able to send ulaw.

If you don't want to start listening until you send the ACK, then don't
send
an SDP in the INVITE.  Wait until the ACK to send it.

Tim

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Maciek Kaminski
Sent: Wednesday, February 11, 2004 12:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Can't connect KPhone to asterisk


Regovich, Timothy wrote:
>Not ACK'ing an invite can be problematic for the statemachine.  Without
the
>ACK, the Dialog is not in  acorrect state.
>
>As for the SDP goes, the KPHONE is offering what it can accept, and
asterisk>is doing the same.  There is no restriction that they must match.  You can
>change your offer in the ACK, or with a re-INVITE.
>  
>Well, they must intersect:
"For streams marked as sendrecv in the answer, the "m=" line MUST
contain at least one codec the answerer is willing to both send and 
receive, from amongst those listed in the offer. The stream MAY indicate 
additional media formats, not listed in the corresponding stream in the 
offer, that the answerer is willing to send or receive (of course, it 
will not be able to send them at this time, since it was not listed in 
the offer).
>As for the immediate transmission : yeah, it does seem a little strange
>doesn't it?  But that is the way that I have seen almost all UAs work. 
The
>implication is that your offer must be a valid, not a conditional offer :
>when you say you accept GSM on port 8000, you better have a listener on 800
>ready to go. 
>  
>Optimistic strategy...

Maciek Kaminski

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



------------------------------------------------------------------------------
Notice:  This e-mail message, together with any attachments, contains
information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New
Jersey, USA 08889), and/or its affiliates (which may be known outside the
United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan as
Banyu) that may be confidential, proprietary copyrighted and/or legally
privileged. It is intended solely for the use of the individual or entity
named on this message.  If you are not the intended recipient, and have
received this message in error, please notify us immediately by reply e-mail
and then delete it from your system.
------------------------------------------------------------------------------

Steve

2004-Feb-12 16:19 UTC

head link

[Asterisk-Users] Can't connect KPhone to asterisk

I had kphone working just fine before I (wiped everything) and installed 
Fedora. If you still need it I canreinstall it and let you know how I 
got it to work.

On Wednesday 11 February 2004 11:42 am, Regovich, Timothy
wrote:> Not ACK'ing an invite can be problematic for the statemachine. 
> Without the ACK, the Dialog is not in  acorrect state.
<snip>

-- 
Steve

__________________________________________________
You actually need to constantly be alert 
 and willing to handle things, or life 
   will find a way to get you good!

asterisk users - Feb 2004 - Can't connect KPhone to asterisk

[Asterisk-Users] Can't connect KPhone to asterisk

[Asterisk-Users] Can't connect KPhone to asterisk

[Asterisk-Users] Can't connect KPhone to asterisk

[Asterisk-Users] Can't connect KPhone to asterisk