[Spce-user] Masked contact parameter 'ngcpct' missing

Daniel Grotti dgrotti at sipwise.com
Mon Jan 20 05:52:32 EST 2014


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Hi Marc,
the problem is the ACK:

New request - M=ACK R=sip:ngcp-lb at 198.51.100.34


the ngcpct has been stripped away.

Since the contact header is :
<sip:ngcp-lb at a.b.c.d:5060;ngcpct=7369703a3132372e302e302e313a35303830>

ACK must contain the RURI equal to the 200OK's contact header.
As Ansrew said, without 'ngcpct' parameter proxy has no way of knowing
where to send an ACK message (asterisk or sems).


Daniel




On 01/20/2014 11:48 AM, Marc Storck wrote:
> Hello Daniel,
> 
> I have attached the requested logs as TXT file.
> 
> Best regards,
> 
> Marc Storck
> 
> 
> 
> 
>> Hi, Since 200 OK contains Contact 
>> <sip:ngcp-lb at a.b.c.d:5060;ngcpct=7369703a3132372e302e302e313a35303830>,
>>
>> 
I'd expect to see the ACK from outside with the same R-URI.
>> Can you paste all the LB and proxy log ?
>> 
>> Daniel
>> 
>> 
>> 
>> On 01/20/2014 11:04 AM, Marc Storck wrote:
>>> Good morning,
>>> 
>>> I have noticed an issue with calls to "voiceboxpass" from a
>>> VoIP peer.
>>> 
>>> In our case, after the call is answered by  the NGCP Asterisk,
>>> kamailio-lb sends a SIP 200 OK to the VoIP peer with the
>>> following contact he
>>> 
>>> Contact:
>>> <sip:ngcp-lb at a.b.c.d:5060;ngcpct=7369703a3132372e302e302e313a35303830>
>>>
>>>
>>> 
The returned SIP ACK does not contain a contact header field at all, and
kamailio-lb logs the line:
>>> 
>>> Masked contact parameter 'ngcpct' missing, trying fallback to
>>> 'sip:sems at 127.0.0.1:5080' - R=sip:ngcp-lb at a.b.c.d
>>> ID=2936446182-25278 at SVIGateway
>>> 
>>> As far as I can see, the ACK is forwarded to the kamailio-proxy
>>> running on port 5062, however no packet is forwarded to the
>>> asterisk server on port 5070.
>>> 
>>> Kamailio-proxy logs:
>>> 
>>> New request - M=ACK R=sip:sems at 127.0.0.1:5080
>>> F=sip:1234567890 at a.b.c.d T=sip:did-of-voiceboxpass at w.x.y.z
>>> IP=a.b.c.d:5060 (127.0.0.1:5060)
>>> ID=2936446182-25278 at SVIGateway
>>> 
>>> As the SIP ACK never reaches the NGCP Asterisk server, the
>>> Audio Stream is terminated after 16 seconds.
>>> 
>>> The missing Contact header in the SIP ACK message does not
>>> cause any issues when calling a subscriber or a conference.
>>> 
>>> Is there something that can be done to fix the access to the
>>> NGCP Asterisk server?
>>> 
>>> Regards
>>> 
>>> Marc Storck
>>> 
>>> 
>>> 
>>> _______________________________________________ Spce-user
>>> mailing list Spce-user at lists.sipwise.com 
>>> http://lists.sipwise.com/listinfo/spce-user
>>> 
>> 
>> _______________________________________________ Spce-user mailing
>> list Spce-user at lists.sipwise.com 
>> http://lists.sipwise.com/listinfo/spce-user
> 
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