practice, there is."
- Original Message -
From: $...@r\/|>r!`/@
To: hanifa.mohammed
Cc: sip-implementors@lists.cs.columbia.edu
Sent: Tuesday, October 12, 2010 4:42 PM
Subject: Re: [Sip-implementors] Contact: * - in 200 OK of REGISTER
Hi,
why does the reg
all,
While de-registering, if REGISTER has Contact as '*', can the Registrar
put '*'
in the Contact header of 200 OK response?
Is this valid?
Best Regards,
Mohammed Hanifa
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Hi all,
Can a FQDN "p1-cscf.open-ims.test" resolve to 3 IP addresses as below?
1. 172.16.8.10 - UDP and no signal security support
2. 172.16.8.20 - UDP, TCP but no signal security support
3. 172.16.8.30 - UDP, TCP and signal security support [TLS]
We feel that all th
Hi all,
Is the below requirement frm RFC3261 applicable for response also? If yes, and
if the VIA
of the response is changed to TCP where as the request has UDP, wont there
be any side effect, while processing this response by the UAC or Proxy Client
txn?
Pl clarify.
"If a request is within
Expires header in INVITE shall be used.
Excerpt from RFC3261:
The UAC MAY add an Expires header field (Section 20.19) to limit the
validity of the invitation. If the time indicated in the Expires header
field is reached and no final answer for the INVITE has been received, the UAC
core SHOULD gen
No CANCEL will be sent to Callee. Simply, Proxy clears the txn with callee.
408 will be sent to Caller.
Best Regards,
Mohammed Hanifa
"In theory, there is no difference between theory and practice. But, in
practice, there is."
- Original Message -
From: $...@r\/|>r!`/@
To: Harbha
all,
I am trying to download SFTF.
Below links are tried but cud not get it.
> http://www.sipfoundry.org/sftf-user-guide.html#downloading, where it
> https://scm.sipfoundry.org/rep/sftf/.
https://lscm.sipfoundry.org/rep/sftf/
Could anyone share any hint on this?
Best Regards,
Mohammed
Hi all,
This is regarding sending VoIP metrics report for SIP session.
In the metrics report, There are two headers , called CallID and DialogID.
Though the header DialogID is to be filled withthe dialog information of the
SIP INVITE session, all the examples contradicts it.
Pl suggest how D
Hi all,This is regarding sending VoIP metrics report for SIP session.In the
metrics report, There are two headers , called CallID and DialogID. Though the
header DialogID is to be filled withthe dialog information of the SIP INVITE
session, all the examples contradicts it.Pl suggest how Dial
all,
Pl find the below snippet in RFC 5628!
* It should remove any temporary GRUUs with a "callid" attribute
value different from that in the value of the "callid"
attribute of the , or with a "cseq" attribute with
value less than the value of the "first-cseq" att
Hi all,
As per the below references of RFC3261,
> 18.1.2 Receiving Responses
>
> When a response is received, the client transport examines the top
> Via header field value. If the value of the "sent-by" parameter in
> that header field value does not correspond to a value that the
> client tr
Pl find the below snippet in RFC 5628!
* It should remove any temporary GRUUs with a "callid" attribute
value different from that in the value of the "callid"
attribute of the , or with a "cseq" attribute with
value less than the value of the "first-cseq" attribut
Hi,
In GRUU, for every REGISTER, the Registrar gives a different temp-gruu
value. i.e. even for each
refresh, a temp gruu is given. Registrar guarantees that all temp gruu's are
valid till the user
deregisters or change-in-callid value.
My question is that whether the first temp-gru
Inaki,> Is '+sip.instance' a Contact URI param or a Contact header param?> The
registrar should store all the Contact URI params and also reply them.It is
a Contact header param. Is there any diff between the processsingof header and
uri parameters at Registrar?
Contact:
;pub-gruu=
all,
In REGISTER request, UA puts '+sip.instance' param in the contact
header.
Isn't it mandatory for Registrar to keep the parameter while sending 200 ok
?
Pl find the related content from RFC5627.
5.2. Generating a REGISTER Response
When generating the 200 (OK) response to the REGISTE
SIP URI
El Jueves, 4 de Marzo de 2010, hanifa.mohammed escribió:
> Hi all,
>
> With respect to tel uri, there are two types:
> 1. Global number - that starts with '+'
> 2. Local number - others
>
> Similarly, is there any thing called Global Sip uri and Loca
Hi all,
With respect to tel uri, there are two types:
1. Global number - that starts with '+'
2. Local number - others
Similarly, is there any thing called Global Sip uri and Local Sip uri.
My assumption:
Suppose the tel uri is converted into sip uri. If the user info part of the
si
Hi,
To update the remote target set, UPDATE/reInvite shall be used.
Assume, reInvite is used.
So, shall UAS update the remote target set, immediately after receiving
the re-INVITE?
Or only if 200 Ok is sent for this reInvite ?
Assume, the sip stack receives the reInvite and intima
"java -jar c07-sip-r2.jar -touri soma_tort...@172.16.8.94 -fromuri
soma_f...@172.16.8.94 -sendto 172.16.8.82 -dport 50025 -showreply -delay 3000
-lport 6070 -teardown"
This is what i used to run this. Hope, this helps.
The destination IP and Port may need to be explicitly specified.
Also, the lo
Hi all,
an excerpt from RFC 5627
"However, a UA MUST be prepared for the public GRUU to change from a
previous one, since the
persistence property is not guaranteed with complete certainty."
So, even for a refresh-REGISTER, the public GRUU may be different. In
that case, for all
existing s
Hi all
Below is an excerpt from RFC5626;
" The sequence of reg-id values does not have to be sequential but MUST
be exactly the same sequence of reg-id values each time the UA
instance power cycles or reboots, so that the reg-id values will
collide with the previously used reg-id values
Hi all,
It is very common that a single UA stack, in a device, shall be used by
many users. In that case,
shall all users use a common instance-id? Will it give any trouble, in real
time?
Pl clarify whether instance-id is an user configuration or UA configuration.
Below is an excerpt from
Hi all,
When the 401 resp of Register has "Supported:outbound", can UAC
assume that the server has outbound support. Or, should it wait for 200 Ok
resp?
Below is an excerpt from RFC 5626.
The UAC examines successful registration responses for the presence
of an outbound option-tag in a
Hi,
On receiving 407 for the intiail INvITE request, should the Route-set be
updated in the caller side?
Pl clarify.
Best Regards,
Mohammed Hanifa
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Hi all,
we are trying to use VoIPER testing tool to test our sip client.
Based on the below doc, we are able to run it and packets are sent across.
http://www.unprotectedhex.com/voiper-wiki/index.php/VoIPER_Usage_Examples#Post_testing_-_Debugging_crashes
http://voiper.wiki.sourceforge.net/VoI
vent header. If the
dialog ID matches,
but Event package is NOT suppoertd, 489 shall be sent from Subscriber also.
Pl clarify.
Best Regards,
Mohammed Hanifa
- Original Message -----
From: Pandurangan R S
To: hanifa.mohammed
Cc: sip-implementors@lists.cs.columbia.edu
Sent: Friday, May 29, 2
Hi,
Also refer Sec "3.3.4. Dialog creation and termination" of same RFC:
If an initial SUBSCRIBE request is not sent on a pre-existing dialog,
the subscriber will wait for a response to the SUBSCRIBE request or a
matching NOTIFY.
Responses are matched to such SUBSCRIBE requests if th
Hi all,
A SUBS req for event 'presence' is sent to Server. Server sends a NOTIFY
with event 'reg'.
Here, the dialog id (callid, from and to tag) matches exactly, but Event is
different.
What error response has to be sent by the client for NOTIFY? 481 or 489
or xxx?
I guess, 481 me
All,
Should Expires header be startd for re-Invite?
Expires header in initial INVItE helps to avoid the race condition of
infinite waiting since transaction
stops timer B, on receiving the first 1xx. The same can happen for reINVITE
also.
If Expires can be started, should CANCEL be
Hi all,
1. When expires header does not have any value, can it be treated as zero?
2. Does any header or header paramter in SIP grammer accept a negative
value?
Pl clarify.
Thanks in adv,
Mohammed Hanifa
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Hi all,
Normally, UAC refreshes at half interval of the expires value. i.e. for
every half an hr if
the expires value in 200 is 1 hr/
When register refresh receives 408,
- should UAC immed terminate all active session such as call,
subscription, ...?
- should retry peri
Hi all,
an excerpt from rfc 3261:
if the response for a request within a dialog is a 481 (Call/Transaction Does
Not Exist) or a 408 (Request Timeout), the UAC SHOULD terminate the dialog. A
UAC SHOULD also terminate a dialog if no response at all is received for the
request (the client transac
For web services, gSOAP libraries are there. This exposes C APIs to
send/receive web service request.
Best Regards,
Mohammed Hanifa
- Original Message -
From:
To:
Sent: Thursday, January 29, 2009 5:19 AM
Subject: [Sip-implementors] Communication between SIP and Web Services
Hello
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