Hi Jānis,
The ACK can have the different request URI if the 200 OK of INVITE contains an
strict router address in Record-Route header. The proxy modifies the
Request-URI of ACK only if the next hope is a strict router.
Regards,
Alok Tiwari
Aricent
-Original Message-
From: sip-implement
Hi Keerthi,
The Max-Forwards header field is used with any SIP method to limit the number
of proxies or gateways that can forward the request
to the next downstream server. Since Registrar/Presence server are acting as
end point for the SIP messages, Max-Forwards header is not mandatory.
Re
emote address from Request URI will be set as strict router in Route
header.
Regards,
Alok Tiwari
Aricent
From: Vivek Gupta [mailto:vivek.gu...@globallogic.com]
Sent: Thursday, November 15, 2012 10:41 AM
To: Pranab Bohra
Cc: Alok 2 Tiwari; Vivek Batra; sip-implementors@lists.cs.columbia.edu
Subjec
Hi Vivek,
In first case, request URI will be set as 'sip:1...@abc.com' and Route header
will have 'sip:proxy2.com' as loose router.
In second case, request URI will be set as 'sip:proxy2.com' and Route header
will have 'sip:1...@abc.com' as strict router.
Regards,
Alok Tiwari
Aricent
-Origin
Hi Tamjid,
As per RFC-3261, Contact is optional in 1xx but it is mandatory in 2xx.
Refer table in section 20.1.
Header field where proxy ACK BYE CAN INV OPT REG
___
Contact R
Hi Antonio,
200 OK of INVITE contains the multiple Record-Route headers. All subsequent
requests from UAC should follow the path as mentioned in Record-Route header
and Address in contact header should be the final destination.
In the mentioned scenario, 200 OK have the following Record-Route
Hi Manolis,
UAS cannot ask for a PRACK without "Require:100rel". Also RSeq is used in 1xx
provisional response only.
Please refer the following sections in RFC-3262.
3 UAS Behavior
The provisional response to be sent reliably is constructed by the
UAS core according to the procedures of S
Hi Pranab,
As per RFC 3261- section 12.2.1.1 Generating the Request
Requests within a dialog MUST contain strictly monotonically
increasing and contiguous CSeq sequence numbers (increasing-by-one)
in each direction (excepting ACK and CANCEL of course, whose numbers
equal the requests bein
Hi Pranab,
As per RFC-3261, section 12.2.1.1,
If the local sequence number is not empty, the value of the local
sequence number MUST be incremented by one, and this value MUST be
placed into the CSeq header field. If the local sequence number is
empty, an initial value MUST be chosen us
Hi Alex,
"npdi" is a parameter in Request URI. So it neither can appear in the user
part nor in the domain part.
It will appear as a uri-parameter as per following grammar.
Request-Line= Method SP Request-URI SP SIP-Version CRLF
Request-URI = SIP-URI / SIPS-
Hi Keerthi,
For all the SIP entities, Basic SIP rules are same.
IMO, In these cases, Proxy and Registrar should send 416 "Unsupported URI
Scheme".
Regards,
Alok Tiwari
Aricent
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists
Hi Manju,
As per grammar defined in RFC-3261, syntax for Call-ID is as follows:
Call-ID = ( "Call-ID" / "i" ) HCOLON callid
callid = word [ "@" word ]
word= 1*(alphanum / "-" / "." / "!" / "%" / "*" /
"_" / "+" / "`" / "'" / "~" /
"("
Hi Rajanagam,
AS per Section: 9.2, RFC-3261,
9.2 Server Behavior
The CANCEL method requests that the TU at the server side cancel a
pending transaction. The TU determines the transaction to be
cancelled by taking the CANCEL request, and then assuming that the
request method is anyth
Hi Tabt,
In SIP, when receiving an INVITE request, receiver sends the response of INVITE
at address received in via header of INVITE. The Contact in INVITE is used by
receiver to send the subsequent request directly to the SIP end point after
call establishment.
Thanks,
Alok Tiwari
Aricent
-
Refer RFC-3261, section-13.2.1:
"If the initial offer is in an INVITE, the answer MUST be in a
reliable non-failure message from UAS back to UAC which is
correlated to that INVITE."
In your scenario, since Terminating SIP does not support 100 Rel and answer SDP
is received in unreliable 180,
Hi Siddhardha,
Please refer the table mentioned in RFC-3261 section-20.
Thanks,
Alok Tiwari
Aricent
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Siddhardha
Garige
Sent: Friday, September 03,
Hi Manoj,
To hold a call, UA should send the SDP in INVITE. Other party can put the call
on hold based on media change in SDP.
If SDP is not present in INVITE, other party must send the SDP in 200 OK as a
new offer and UA should send the answer SDP in ACK.
Thanks,
Alok Tiwari
Aricent
-Orig
Hi Mia,
IMO, it would be better to implement a new timer to handle such scenario. This
timer should be stopped if any final response of Re-INVITE is received. In case
of timer expiry, terminate the dialog. The timer duration should be greater
than Timer-B.
Thanks,
Alok Tiwari
Aricent
-Orig
Hi Michael,
Static codec: Static codecs have the defined payload number for a particular
codec. Therefore, the use of a=rtpmap attribute for static payloads is optional.
Dynamic codec: Dynamic payload types are in range of 96-127. Payloads in this
range do not refer to a particular codec; inste
Hi Nahum,
The REGISTER request submitted after receiving the 401 response should contain
the "Authorization" header with authentication credentials. Sever will
authorize the REGISTER request based on the authentication credentials received
in "Authorization" header.
Please refer RFC-3261 sect
Hi Chozhan,
SIPS allows resources to specify that they should be reached securely. It
mandates that each hop over which the request is forwarded up to the target
domain must be secured with TLS. Therefore, SIPS uri scheme can used only in
case of TLS but SIP URI can be used in any transport mod
Hi Amar,
A dialog is a peer-to-peer SIP relationship between two user agents that
persists for some time. The dialog facilitates sequencing of messages and
proper routing of requests between the user agents. Adialog is identified by a
call identifier, local tag, and a remote tag.
A session is
Hi Elison,
The absence of an Allow header field MUST NOT be interpreted to mean that the
UA sending the message supports no methods. Rather, it implies that the UA is
not providing any information on what methods it supports.
Refer RFC-3261 section: 20.5
Thanks,
Alok Tiwari
Aricent
-Origi
Hi,
Proxy can challenge the UA by sending the '407 Proxy Authenticate Response'.
Please refer the following section in RFC-3261.
21.4.8 407 Proxy Authentication Required
This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with
Hi Antoine,
The BYE request contains the two semicolon (;) before the To-tag.
Retry the scenario after removing one extra semicolon before To-tag.
Thanks,
Alok Tiwari
Aricent
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.c
Hi Josep,
When a request is challenged with 407 (containing proxy-authenticate header),
the request should be re-submitted with proxy-authorization header.
Please refer the following section is RFC-3261:
22.3 Proxy-to-User Authentication
Similarly, when a UAC sends a request to a prox
Hi Zabi,
INVITE and Re-INVITE are different transactions. 200 OK for INVITE and
Re-INVITE will have the different CSeq value and Branch parameter in Via header.
UA should send the same ACK for INVITE after receiving 200OK.
Thanks,
Alok Tiwari
Aricent
-Original Message-
From: sip-implem
Hi Vivek,
Outbound proxy is only used while making an outgoing call.
In case of incoming call, the responses are sent on via and after call is
established, the SIP messages should be send to SIP contact or record-route.
Thanks,
Alok
Aricent
-Original Message-
From: sip-implementors-bou
Hi Manoj,
Via is used to route responses whereas Route and Record-Route headers are used
to route requests. User agents use Record-Route headers to build Route headers.
The other routing related headers are Path header (Refer RFC-3327) and
service-route header (Refer RFC-3608).
Thanks,
Alok Ti
Hi Anna,
The ACK should be ignored. An ACK is only sent in response to a response to an
INVITE request.
Thanks,
Alok Tiwari
Aricent
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Gellatly,
Ann
Hi Sharan,
ACK is the separate transaction only when INVITE is responded with 2xx response.
Please refer the section-17 in RFC-3261.
17 Transactions
SIP is a transactional protocol: interactions between components take
place in a series of independent message exchanges. Specifically, a
Hi Jon,
Requests other than CANCEL can receive a 3xx response.
Please refer the "section 8.3 Redirect Servers" in RFC-3261:
"A redirect server does not issue any SIP requests of its own. After
receiving a request other than CANCEL, the server either refuses the
request or gathers th
Hi,
As per spec 3GPP TS 24.229 V8.0.0 section: 5.1.1.5.1
Authentication is performed during initial registration. A UE can be
re-authenticated during subsequent reregistrations, deregistrations or
registrations of additional public user identities. When the network requires
authentication or r
Hi,
As per RFC-3261, the retry after timer is started by UAC itself after deciding
the retry after duration based on call id ownership. So there is no need of
retry after header in 491 response code. But since your implementation is as
below:
I am adding the party name to make you understand t
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