From: Sarkar, Uttam [EMAIL PROTECTED]
If the message INVITE [EMAIL PROTECTED] SIP/2.0 received at
uaIPAddress, then it would be the decision of UA to reject or accept.
If the UA is flexible enough then it can accept the Request otherwise it
can reject by sending 404 Not Found.
From: Rishabh Garg [EMAIL PROTECTED]
Ours is a B2BUA. Currently if we are receiving the 4xx response
(for eg. 486 Busy Here) from the terminating called party,then
after ACKing the same we are sending the BYE to originating calling
party and dropping the connection instead of
From: Ivar [EMAIL PROTECTED]
How many implementations really do it or consider doing it so ?
http://bugs.sipit.net/show_bug.cgi?id=769
The sipX stack does so. I expect that any stack that is deployed in
the real world does so.
Dale
___
We're noticing that when the SIP network gets congested, phones will
be fairly frantic about resending requests that they do not receive
(provisional or final) responses for. Unfortunately, this only
increases the load on the proxy, which does not help the situation.
For INVITEs, the proxy sends
From: Nasir Khan [EMAIL PROTECTED]
Is the fix defined somewhere?
(This is discussed every few months on this list.)
The way to fix the problem is to distinguish between the state of the
*dialog* and the state of the *transactions* that happen within the
dialog.
Once a request is sent or
From: Naresh R [EMAIL PROTECTED]
+ Is there any website or document that has given any simpler combination
about what all error responses that can be expected for every SIP request?
+ Is it Valid if I assume that every SIP method can expect any SIP response
(2XX to 6XX) unless and
From: Stephan Steiner [EMAIL PROTECTED]
Shouldn't the reply contain the following Contact header for full RFC 3261
compliance:
Contact: sip:[EMAIL PROTECTED];transport=UDP;expires=360
(I added the brackets because the transport requires a semicolon in the URI
and then
From: Rami Eitan [EMAIL PROTECTED]
My question is does anyone know of or has ever come across a
request message without the VIA header in it?
If there was no Via header, the recipient would have no way to know
where to send the response.
Dale
From: George AK [EMAIL PROTECTED]
Is it a must to have offer-answer (SDP based) in an INVITE session?
Can I have a INVITE-200-ACK with out an SDP Offer-Answer?
RFC 3261 explicitly allows that there might be other ways to describe
an offer or answer than SDP, and so that media-type may
From: Hagai Sela (TA) [EMAIL PROTECTED]
Doesn't the second part of the paragraph contradict the first part? Why
should the receiving side's implementation accept the large packet if
the sender is not supposed to send it?
Be strict in what you send; be liberal in what you accept.
From: Zeev Kamelmacher [EMAIL PROTECTED]
I'm trying to implement a validation of the sip syntax. I need some
clarifications concerning the SIP BNF (rfc 3261):
1. In case of parsing the request-line (Request-Line = Method
SP Request-URI SP SIP-Version CRLF)
From: Hannes Tschofenig [EMAIL PROTECTED]
no matter how you call it but there will always be a (VoIP) application
service provider involved in the emergency services case.
I don't see that this is guaranteed. E.g., a VoIP mobile phone could
connect to an open WiFi hub at a coffee shop
From: Sumin Seo [EMAIL PROTECTED]
What is a RECOMMENDED UAS behavior? sending final response to
INVITE or sending BYE?
Either is acceptable, but by sending a non-success response to INVITE,
the UA can more accurately specify what the problem is. It will also
most likely prevent billing
From: Attila Sipos [EMAIL PROTECTED]
Looking on an ethernet snooper, the beginning of
the display-name is:
22 cc e1
Now obviously 22 is the open quote ( ).
Next is cc.
Now, to me, cc is UTF8-NONASCII so next comes 1 UTF8-CONT.
But no, the next byte is e1 which is not a
Ideally, a SIP UA should never register its contact address for longer
than the length of time that it knows the contact address will be
valid. If the UA's address comes from DHCP, this would seem to
suggest that the UA should never ask for a longer registration than
the remaining length of its
From: Sweeney, Andrew \(Andrew\) [EMAIL PROTECTED]
I am trying to determine if transport=tcp must be added to a request
when the user is going to run over TCP.
First, make sure you've read RFC 3263. The general philosphy is A
SIP agents has a SIP message that it needs to send to a
From: Michael Procter [EMAIL PROTECTED]
I think that 'no circumstances' is a little strong. From RFC3261
Section 18.2.1:
Ouch! I certainly stand corrected there. It shows that I haven't
worked with NATed systems enough. The full precedures are in 18.2.2,
but they depend on the
From: Zarko Coklin [EMAIL PROTECTED]
1. Can Registrar have liberty to change realm in 407 between
REGISTER requests?
2. How should UA react if it gets different realms for 2 different
REGISTERs? Some UA cached REALM1 and used it all the time not
respecting that Registrar
From: [EMAIL PROTECTED]
When adding the XML body to a SIP message, should each line in the
XML body be terminated by a \r\n sequence or only a \n
character?
Lines in an XML document can be ended with \n, or \r\n, or even by
\r, but all lines must be ended the same way. All line
From: venkatesh chandran [EMAIL PROTECTED]
I have got the following information regarding message size details while
googling...
The maximum length of each part of a message is shown in Table 5.1-1. The
length of the whole request/response messages
Note that this is
From: Gayathri Madda [EMAIL PROTECTED]
can u please suggest how to parse this as per ABNF Rule
sip:5550100;phone-context=+1-630;tgrp=TG-1;[EMAIL
PROTECTED];user=phone
As per RFC we use @ for parsing host and user part
Here in this case :
From: Jagan Mohan Reddy S [EMAIL PROTECTED]
What is the behavior of UAS on invalid sent-by field in the
incoming request?
It is probably unwise for a UAS to respond to a message that is so
damaged that the message does not clearly specify the way it is to be
responded to. But since any
From: Ira Kadin [EMAIL PROTECTED]
Could you provide me with some number of different SIP stacks
performance (reciprocate, Rad Vision, ...)
Could you suggest the characteristics we can use to measure the SIP
stack performance (for example - number of simultaneous calls, call
From: Michael Hirschbichler [EMAIL PROTECTED]
I was wondering, what should be or is the default behaviour of a
Proxy/UA when it receives an unknown header, like S-something: somewhere.
I guess, the Proxy ignores this header and forwards the SIP message to
the callee including
From: Jagan Mohan Reddy S [EMAIL PROTECTED]
Is there any limit for the size of SIP header?
No.
PROTOS is sending an INVITE with too many (200 bytes) of junk
characters in VIA header. Can we treat this message as malformed
message and drop the packet without doing further
From: Jeroen van Bemmel [EMAIL PROTECTED]
Which is correct:
a) 200 OK for deregister with Contact header with expires=0 in it
b) 200 OK for deregister without any Contacts
A better way to think of it is What does a 200 for REGISTER mean?
It must list all registered contacts, with
From: Franz Fischer [EMAIL PROTECTED]
Could one of you commit that this is a bug in x-lite ?
In my opinion the expiration in the response and the notify request
should be 10 or the client should respond 423 interval too brief
RFC3265 3.1.1 says The period of time in the response
From: J Jayakumar [EMAIL PROTECTED]
In RFC 3261 under section 10.2.4 it has been specified that the
200 ok response from a registrar may or may not have a expires parameter in
the contact header. And in Section 10.3. 8 it has been said that the 200 ok
from a registrar MUST have a
From: Reynolds, Paul [EMAIL PROTECTED]
D) Send a failure (and if so, what? 400?)
You should send a failure (you quoted the text that requires you to do
so). The failure you should send is 500 (which is used for all
situations where CSeq is l.e. a previously seen CSeq), officially
called
From: [EMAIL PROTECTED]
In case of DTMF digits, a single RFC 4733 packet carries a single DTMF
digit in my understanding. Triple redundancy is the basic means to target
RTP packet losses, i.e. the receiver must receive at least one out of
three.
Be careful -- An RFC 4733 packet
From: Anders Kristensen [EMAIL PROTECTED]
Paul Kyzivat wrote:
I'm not certain it would even get that far. If it has sent a 1xx with
100rel and it doesn't receive a PRACK in response, then I think it will
abandon the call at that point. (Though the proper way to abandon the
From: chom sri [EMAIL PROTECTED]
Can we send a 305 response from a redirect server or they should
only be generated by useragents?? Similarly can an ordinary
redirect server generate 380 response??
According to RFC 3261, 305 must only be generated by UASs, by which I
assume it means,
From: johnny kao [EMAIL PROTECTED]
1 . RFC3261 says on page 77: Independent of the method, if a request
outside of a dialog generates a non-2xx final response, any early
dialogs created through provisional responses to that request are
terminated.
Does it only describe the
From: Mayank Kamthan [EMAIL PROTECTED]
The RFC 3841, in the section 5 titled 'UAC Behavior' says,
The Accept-Contact, Reject-Contact, and Request-Disposition header
fields in an ACK for a non-2xx final response, or in a CANCEL
request, MUST be equal to the values in the original
From: Paul Kyzivat [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
From: Rayees Khan [EMAIL PROTECTED]
It is not always the case. In case there are offer-answer exchanges with
PRACK and UPDATE before 200 OK is sent, having SDP in 200 OK same as 183
is not a good
From: Paul Kyzivat [EMAIL PROTECTED]
I assume that the obligation of the UAS to put the SDP in the 200 is
removed when it receives a PRACK of a provisional response that had
SDP? That is, if it *doesn't* get PRACKs that it expects, it must put
the SDP in the 200.
I'm not
From: Mushtaq Ilyas [EMAIL PROTECTED]
Is a Cancel request (meant to cancel an invite request) part of the
Invite transaction?
No, the CANCEL has its own transaction (and thus gets its own
response).
Dale
___
Sip-implementors mailing list
From: varun [EMAIL PROTECTED]
What is the use of expires header field in an Invite
request? Will the request get timed out after this
time in case no final response is received?
It means that the invitation is only valid for the specified length of
time.
13.3.1 Processing of the
From: Franz Fischer [EMAIL PROTECTED]
Could one of you have a look at the following sniffered
communication? The Notification fails and I have absolutely no idea
why. I'm trying to establish a communicaton between the sip client
x-lite and an own application. My own application is
From: Ivar [EMAIL PROTECTED]
Sender: [EMAIL PROTECTED]
Hi,
This is dumb question but, if proxy has route header and request-URI is
the proxy local, must proxy forward request or handle it ?
You must describe the situation more carefully. A proxy receives a
request. In the
From: venkatesh chandran [EMAIL PROTECTED]
In the above case, if no REQUEST-DEPOSITION header is present, will the
INVITE will be recursed or 302 will be proxied back to the UA.Is any RFC
specifying the action?
Either alternative might happen. In particular, the UA must be
prepared
From: Barman, Sibon B \(Sibon\) [EMAIL PROTECTED]
Is changing codec sequence in the 200 OK from 183 legal? I am guessing
it is --- just need to verify if that's the case.
I doubt that it is OK. In any one transaction, the SDP sent by one
agent must always be the same. So you can't
From: Sarkar, Uttam [EMAIL PROTECTED]
When UAC is request for some codec in the INVITE and none of them is
supported by UAS then UAC will get 415 response.
Unfortunately, that is not correct. See RFC 3261 section 21.4.13:
21.4.13 415 Unsupported Media Type
The server is
All server transactions linger for the 3 minutes
Hmm can you point me such place in rfc ?
If i look rfc 3261 17.2.x
Server transactions will destroy too if final response, only in some
cases linger for 4 seconds.
There is an error in 3261 in that it does not clearly separate the
From: [EMAIL PROTECTED]
400 Bad Request might not be appropriate as Basic is syntactically
correct.
Good point.
I think 401/407 is the appropriate option.
Well, there is the tricky special case where the authentication
credentials are Basic (and thus useless), but the request does
From: Bob Penfield [EMAIL PROTECTED]
How about 403 Forbidden?
That seems like a poor choice to me. Yes, the UA shouldn't have sent
it, but there's a general principle regarding credentials: the
supplicant might present you with a pile of credentials, and it's your
job to sort through them
From: Ivar [EMAIL PROTECTED]
But what happens to server transaction after cancel ?
Logical is that it will be terminated and disposed (because nothing to
do with that server transaction), but can't see place what describes it.
Client transaction can't dispose at once, thats
From: Mushtaq Ilyas [EMAIL PROTECTED]
So that means that I (Proxy Server) will never get a request
(containing authorization header) from a client that I have not
challenged before?
That is true, if the client is behaving correctly.
But if the server was restarted, it may not
From: Zarko Coklin [EMAIL PROTECTED]
I have a practical question. SIP device subscribes to
a SIP server, notifications are sent for some time and
then server reboots. When this happens CSeq on server
side is reset to 1.
Next time SIP phone subscribes, server accepts a
From: Chaney, Charles \(SNL US\) [EMAIL PROTECTED]
I am trying to determine how a forking proxy (and UAC for that matter)
should handle a UAS 1xx response and another UAS responding with a 3xx.
I'm unable to find a definitive answer in RFC3261. While 3xx responses
are typically
From: Daniel Corbe [EMAIL PROTECTED]
I'm seeing the following behavior out of one of my endpoints, Its
trying to hang up a call which doesn't exist so the natural thing to
do seems to be to send a 481, like so:
--- BYE
-- 481
-- BYE
-- 481
-- BYE
-- 481
--
In case of BBUA, is it possible for proxy to generate 1xx
response for anINVITE, before receiving any reply from
terminating SIP client.For Ex. SIP client A calling another
SIP client B, and before receivingany reply from the SIP client B
for the initial INVITE, can
From: Thirumal Margabandhu [EMAIL PROTECTED]
Recently i tested one sip client application, there i can find , the
from and to field are same.
From: sip:[EMAIL
PROTECTED];tag=e02335237cb143fe9729ed3955f68d24;epid=56747ddc83
To: sip:[EMAIL PROTECTED]
Can any one answere, why
From: Adarsh Guler [EMAIL PROTECTED]
Because by default it will be sending REGISTER messages to UDP
Port. Then how to inform UAC about the TCP Connection and TCP Port
It can be configured to know. Or it can determine the transport to
use by looking at the SRV records for the domain
From: Nina Garaca [EMAIL PROTECTED]
I have a question about terminating the dialog during the session
modifiaction with a reINVITE:
Q: Does this claim also refer to reINVITE ?
/ /
/ RFC 3261/15 /
/ The caller's UA MAY send a BYE for either/
/ confirmed or early
From: Nina Garaca [EMAIL PROTECTED]
A B
INVITE
|--|
180 (with To tag)
|--| /Early dialog established/
UPDATE
|--|
408/481
From: Bu, Wenfei \(Leo\) [EMAIL PROTECTED]
UAC UAS
---INVITE w/o SDP
---100
---180 w/ Require: 100rel SDP
---180 w/ Require: 100rel SDP
---180 w/ Require: 100rel SDP
---180 w/
From: Steve Langstaff [EMAIL PROTECTED]
In the following trace (hopefully anonymised) I can't see how endpoint B
can expect to receive or send audio, since it appears to negotiate away
all but the telephone-event (101) codec.
It's clearly an error (in the practical sense) on the part
From: Tang Xi [EMAIL PROTECTED]
1) UAC sends a REFER to UAS
2) UAS sends a 202 response back and do the reference, but doesn't send
NOTIFY with 100 trying
3) After some seconds the UAC terminates the REFER application, cause it's
timer out
Is the behavior of UAC correct?
From: [EMAIL PROTECTED]
Why this peer-to-peer relationship is not considered as a dialog?
The basic answer is Because RFC 3261 says it isn't.
A dialog has a call-id, to-tag, etc. The creation of a dialog sets
the contacts and route-set. But the big difference is *functional*.
That data
From: Subbarayalu Subbiah [EMAIL PROTECTED]
i want to know ,is it possible for proxy 1 to by pass the request
(since it is not processing these header) to proxy 2 ( where the actual
processing is done for this header) instead of sending a 420 response.
No, if there is a
From: Einat Soudry [EMAIL PROTECTED]
Is it a normal behavior that the port in the Record -Route/Via
header is 5060 although it is different from the used source port?
It's perfectly legal, though not that common. E.g., the sipX
open-source proxy does this. But agents that do not use
From: Barman, Sibon B \(Sibon\) [EMAIL PROTECTED]
One thing I am still not clear if 3261 specifies clearly that the
contact address in the response has to be the same (in literal sense) as
the contact address in the registration request. Is the public address
corresponding to a
From: Nina Garaca [EMAIL PROTECTED]
According to RFC 3261, RFC 3428 and RFC 2976, INFO, OPTIONS, MESSAGE
requests are mid-dialog requests. Does it mean that these can be sent or
received during the early dialog also?
In addition to what others have said:
OPTIONS should clearly be
Date: Wed, 14 Mar 2007 11:23:53 +0100
We recently ran into the following interop issue: client receives a NOTIFY
out-of-order and sends a 500 error response, presence server terminates the
subscription
A very good point!
Another argument that CSeq out of order should have its own
From: Barman, Sibon B \(Sibon\) [EMAIL PROTECTED]
When a user agent sends a REGISTER request with private address in the
Contact header, the session border element sends a response with the
UA's public IP address in the contact header as well as in the Via
header's rport and
From: Bin Chen [EMAIL PROTECTED]
Pardon, what is SBC and SBE?
SBC stands for session border controller. SBE stands for session
border element, which is the term the Sibon Barman used. The two
terms have about the same meaning, but SBE is a more sensible way of
describing it.
Dale
From: Paul Kyzivat [EMAIL PROTECTED]
If the SBC is going to translate the Contact in the request, then it
also ought to back translate the Contact in the response.
It is a bit much to expect UAs to implement workarounds MITM attacks by
SBCs.
Not to mention that such UAs make
From: Michel Eilat [EMAIL PROTECTED]
I would like to know if there are any special
requirement/specs/drafts for SIP trunk (use instead of old TDM
trunk)?
The SIP Forum has a specification for trunking called SIPconnect.
Dale
___
From: NandaKishoreE 71062 [EMAIL PROTECTED]
The parameter will indicate which A/S in the cluster is handling that dialog.
This parameter will be used by a router to route in-dialog messages
to the corresponding A/S.
The problem comes when the A/S has crashed and the router needs
From: Sunil Kumar Verma [EMAIL PROTECTED]
In case of BBUA, is it possible for proxy to generate 1xx response
for an INVITE, before receiving any reply from terminating SIP
client.
For Ex. SIP client A calling another SIP client B, and before
receiving any reply from the SIP
From: Bhanuprasad K S [EMAIL PROTECTED]
I (UAC) sent a REGISTER request to registrar, registrar responded with =
200 OK.
The thing is, in 200 OK response to REGISTER To-Tag is not there, rest =
is fine.
How my UAC should behave upon receiving this 200 OK.
Is UAC is
From: Arif [EMAIL PROTECTED]
Just wanted to clarify my confustion.
I suggest you read section 10 of RFC 3261 carefully.
Dale
___
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
From: SiM [EMAIL PROTECTED]
1.) What i understand is that one way is to use the Dialog event
package and then send a REFER request with the method=CANCEL,
adding the Target-Dialog header, with whatever information is
available from the NOTIFY. i.e a Half Dialog information,
From: Bharrat, Shaun [EMAIL PROTECTED]
Can someone more familiar with this situation comment on whether
it is fact the expected behavior for such a UAC with many AORs?
Appreciate any info or links to prior email threads.
After my misconceptions of the situation were corrected, and
From: Raghu Thodime [EMAIL PROTECTED]
Can anybody explain if following two scenarios are valid and where:
1. UAC receives 407 for the request it sent out and again
it gets 401 for the request it sent out in response to previous
407
2. UAC receives 401 for
From: Chaney, Charles \(SNL US\) [EMAIL PROTECTED]
Should the other contact binding reflect the remaining
(time-to-live) expiration value or be updated to reflect the new
registration (or refresh), i.e., all are reported with the same
expires value. I cannot point to any definitive
From: Yong Xin [EMAIL PROTECTED]
The RFC 4412 (Communication Resource Priority for SIP) does not clearly
state which error code should be used in the case when a particular
namespace has appeared more than once in the same SIP message.
Any suggestion?
400 Bad Request
Dale
From: Joegen E. Baclor [EMAIL PROTECTED]
Which beg the question, is third party subscription allowed by the RFC
in the same manner it is allowed for registrations?
There is no mechanism for it, because when SUBSCRIBE creates a
subscription, the NOTIFYs are sent using the route set and
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
You have my vote, Dale. This leaves me to wonder why SUBSCRIBE, the
only dialog creating request after INVITE leaves such a horrendous hole
in interpretation. I have a registrar/proxy implementation that
receives a gazillion NOTIFYs a
From: Satyendra Tiwari [EMAIL PROTECTED]
I have seen two differnet approaches to solve this:
1. Send a re-Invite with 0.0.0.0 as the IP address in the sdp data
2. Send a re-Invite with the parameter a=sendonly set in the sdp data
caller a=sendonly, callee a=recvonly(response)
From: Paul Kyzivat [EMAIL PROTECTED]
If the NOTIFY arrives out of dialog then I don't think 481 is a suitable
response, unless we also want to overload it further to mean this
should have been in a dialog.
Well, I take 481 to mean This request presupposes the existence of a
dialog,
From: Anders Kristensen [EMAIL PROTECTED]
There are legitimate usecases for SDP with no m= lines, see RFC 3725.
Interesting!
Though I see that RFC 3725 section 9.2 runs into a similar question to
the one that started this thread:
Therefore, it sends an INVITE (1) with SDP that
From: Bin Chen [EMAIL PROTECTED]
I searched some materials about the SIP parser writing, someone
mentioned the SIP spec grammer is a ABNF, but not BNF and it's hard to
use yacc to parse it. Is it true and I want to know why?
I know yacc can parse LALR grammer and does it has
From: erol turac [EMAIL PROTECTED]
If the registrar allows UACs to register with same pin, password
and URI with different IP, the only solution will be that the
incoming INVITE from the UAC5 should be forked to both UAC1 and
UAC2. Forking should be implemented by proxy,
That is
From: Yong Xin [EMAIL PROTECTED]
We have different interpretation regarding to the c= line definition in
RFC 2327.
Session description
c=* (connection information - not required if included in all media)
Media description
c=* (connection
From: erol turac [EMAIL PROTECTED]
If an endpoint sends initial invites with ip address A, and then it sends
sub register messages with a different ip address B,
How does the UAS handle this request ? Does UAS response with 200 OK or 5xx
server error ?
This is the correct way to
From: Bob Penfield [EMAIL PROTECTED]
This is not correct. If the two contact addresses are different, there will
still be two bindings for the address-of-record. The fact that the two
REGISTER messages have the same Call-ID does not cause the second REGISTER
to replace the
From: [EMAIL PROTECTED]
I just ran some tests with various brands of phones and none send
ptime in their SDP, which makes this harder to understand. Here is
one from a polycom. Are you saying that even though 3 media formats
are listed, they are represented as one media stream? Is
From: Vick, Steven [EMAIL PROTECTED]
[sv] the snom is making the call to the sip gw using g729a20, and
the call is accepted.
What SDP is the Snom sending to the GW? What SDP is the GW responding
to the Snom?
If you compare these to the SDP offer/answer when the call is made in
the
From: Vick, Steven [EMAIL PROTECTED]
There in lies the problem. The Aspect endpoint is sending two media
streams.
There are two media streams being offered, but the Ptime is only
represented for the first one on the list. This happens regardless of
how many media streams (or
From: Einat Soudry [EMAIL PROTECTED]
In case there isn't Contact header in REGISTER request (the rfc
says Contact header MAY be included) what should be the Transport
protocol used for incoming requests to that UA? Should it be as
the transport protocol in the Request?
If there
From: Vick, Steven [EMAIL PROTECTED]
My Snom phone only allows 1 sample size setting per codec, and it's
configured for 20ms. The endpoint calling the snom is configured for 10
and 20ms, where 10 will be listed first in the SDP.
It looks like the Aspect endpoint is offering one
I've submitted an Internet-Draft giving guidelines for UAs that want
to implement the dialog event package. In our product's environment
(a true SIP, proxy-based PBX), we've found that to do call control
well, the UAs need to implement the dialog event package well. This
I-D describes the needed
I've updated an Internet-Draft giving condensed instructions for UA
implementors on how to implement GRUU support. (This is updated for
version -11 of the GRUU I-D, but it only discusses public GRUUs,
which is what we need for our environment.) In our product's
environment (a true SIP,
From: Sweeney, Andrew \(Andrew\) [EMAIL PROTECTED]
Why is Fake Forking OK to do?
Because the UAC cannot distinguish it from the situation where two
UASs have received forks of the INVITE and each UAS is establishing a
separate early dialog with the UAC.
What is the issue that the SDP
From: Gayathri Madda [EMAIL PROTECTED]
What is the significance of Branch=0 in via Header
and how does it effects.
The branch value is supposed to be different for every request sent on
by every proxy. So the value 0 is probably a mistake.
Dale
From: Marco Ambu [EMAIL PROTECTED]
implementing a registrar server we found some problems with this part of
RFC 3261 - 10.3 (page 66):
For each address, the registrar then searches the list of
current bindings using the URI comparison rules. If the
binding does not exist, it is
From: Menon.V, Hari [EMAIL PROTECTED]
We were planning to implement a SIP Stack for our company.
The Stack has to support all the SIP functionalities.( Proxy, Registrar,
Presence , SIP User Agent) The stack has to be compliant with RFC 3261
and also some of the extensions like
From: Hagai Sela \(TA\) [EMAIL PROTECTED]
I am looking for a free proxy or a UA which supports RFC 4235
(dialog events). Does anybody know one?
Only user agents generate dialog events, because only user agents
terminate dialogs.
I know that the internal user agents (such as the Park
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