On 22 Oct 2014, at 18:43, Pranav Damele wrote:
> I guess that would make subsequent answer a problem because of session
> level c line 0.0.0.0 ... stream level is good enough.
>
> Regards
> Pranav
> On 22 Oct 2014 22:08, "Paul Kyzivat" wrote:
>
>> On 10
!
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therefore
> duplication is not reasonable.
>
Multiple Diversion headers are definitely allowed:
http://tools.ietf.org/html/rfc5806#section-6.5
So, the fact they look the same doesn't imply that there is something wrong,
IMHO.
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made some fixes so you may
need to get the latest version from trunk.
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digram does not show it,
> right?
>
> I mean (1) is legimate.
>
Indeed, you are right, my memories were a bit rusty ;-) That's actually how
OpenSIPS behaves IIRC and what makes more sense to me, since you avoid
subscribing again.
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you may need to modify 3 documents just for adding a contact: pres-rules,
resource-lists and rls-services, and there is no way to do that operation
atomically.
> if client doing this way, then no need to for the proxy server to send
> the pending state, the server just need
here) 102 would subscribe to 101 again, because she sees
>> that 101 just subscribed to him, he is her buddy, and she has no
>> subscription towards him.
>>
>>
>
> Yes, that would be a good solution if client people agree to do it.
>
FWI
102 did not remove 101 from his contact list.
>
>
> (6) are (1)-(5) above correct behavior based on the sip SIMPLE RFCs?
>
> If so, how to solve the issue?
>
> Is there any standard/RFC way to notify to 102 that 101 allow 102
> again thus cause 102 to re-subscribe 101 again?
>
>
> Just FYI, there was a discussion on it on the sip-router mailing list
> http://lists.sip-router.org/pipermail/sr-users/2012-June/073759.html
>
>
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y the SIP protocol? That would break
interoperability with standard SIP clients.
Anyway, if want to go that route I'd get one of the many available Open Source
SIP stacks (PJSIP, Sofia, reSIProcate, ...) and modify it.
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imer A controls request
retransmissions, and in case of an unreliable transport it defaults to T1.
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condition in how the server handles requests / responses.
OpenSIPS, for example, can relay the response and processes it *later* so an
ACK could sneak in between. I think this case should be handled. OpenSIPS does
handle it, though I'm not skilled enough to give you details on ho
f 180 is
agreed, the registrar will expire that registration binding after exactly 180
seconds, It's up to the device to refresh the registration before it expires.
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Sip
mple2.com, both example.com and
example2.com resolve to 1.2.3.4, so if I call alice@1.2.3.4, which credentials
should I check? al...@example.com or al...@example2.com?
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could not find
any reference to where/if its standardized :-S
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t doesn't matter if a device is not
registered, if its credentials are OK it should be able to make call. If
device2 is online it should get the INVITE else you can reply with 404 for
example.
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Hi,
On Jul 7, 2011, at 5:06 PM, Worley, Dale R (Dale) wrote:
>
> From: sip-implementors-boun...@lists.cs.columbia.edu
> [sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Saúl Ibarra
> Corretgé [s...@ag-projects.com]
>
>
gs at the same time, and I may want to transfer
Alice to Carol, but I'd like to suggest Alice that she uses audio in the INVITE
she sends to Carol.
> I think there is a fruitful area for investigation here.
>
I'd be willing to collaborate if there is interest on the subject :-)
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On Jul 7, 2011, at 11:48 AM, Iñaki Baz Castillo wrote:
> 2011/7/7 Saúl Ibarra Corretgé :
>> When doing a call transfer, the party sending the REFER communicates the URI
>> the recipient is supposed to call by using the Refer-To header. This header
>> may also contain
cate *what* streams should the
recipient use when sending the outbound INVITE?
This would let the REFER recipient know if he should offer audio, video, MSRP
chat or any other kind of stream. Is there any RFC/draft defining this?
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A
to account.
>
Did you check c-ares (http://c-ares.haxx.se/)? I haven't used it myself but
I've seen it in some networking related projects. Just curious.
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apping acc1's PSTN number to acc2. Then you
can query it on your proxy and redirect calls accordingly.
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it detected. It
uses an integer which is mapped to a more comprehensive string in the code,
like "Symmetric", "Full Cone", etc.
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Sip-implemen
i parameters.
>>
>> http://www.iana.org/assignments/tel-uri-parameters/tel-uri-parameters.xml
>
> Life would be better if there was no TEL URI within SIP protocol.
>
>
According to this grammar, the semicolon is not a delimiter for 'user '
parameters, its just a value wh
On 05/10/2011 05:14 PM, Iñaki Baz Castillo wrote:
> 2011/5/10 Saúl Ibarra Corretgé:
>>>> What do you mean? Inbound or outbound relay? Where is the connection
>>>> broken?
>>>
>>> Humm, both cases. This is:
>>>
>>> alice ---
On 05/10/2011 04:02 PM, Iñaki Baz Castillo wrote:
> 2011/5/10 Saúl Ibarra Corretgé:
>> On 05/10/2011 03:42 PM, Iñaki Baz Castillo wrote:
>>>
>>> 2011/5/10 Saúl Ibarra Corretgé:
>>>>
>>>> Session needs to be reestablished IMHO. You don&
On 05/10/2011 03:42 PM, Iñaki Baz Castillo wrote:
> 2011/5/10 Saúl Ibarra Corretgé:
>> Session needs to be reestablished IMHO. You don't know what happened
>> and if its ok to reinvite.
>
> And what about if the MSRP session is establishe through a MSRP relay?
>
On Tue, May 10, 2011 at 10:43 AM, Iñaki Baz Castillo wrote:
> Hi, I'm startign with MSRP and would like to clarify some basic
> concepts. Let me start with this easy question:
>
> - Alice has started a MSRP session with Bob over TCP or TLS.
> - Later, by any reason, Bob closes such TCP connection.
Hi Nancy,
On Mon, May 2, 2011 at 9:38 PM, Nancy Greene wrote:
> MSRP defines Failure-Reports - section 7.1.4 says you can send a
> Failure-Report if an error is detected after you have already sent the 200 OK
> to the MSRP SEND.
>
> Nancy
>
Thanks for the comment. The problem with Failure-Repo
> This is like when you send a mail. The SMTP protocol just asserts you
> that the mail has been received in the destination server, but you
> have no way to determine if it has been delivered to the final user
> (let's forget mail-receipt-configuration features, who uses that?).
>
There is no con
Hi all,
While dealing with file transfer in MSRP the following case happened
to me and I wonder if anyone run into it before or had any thoughts to
share.
Lets say we transfer file test.iso from Alice to Bob. Bob will
acknowledge the reception of every chunk of data just fine. Since I/O
operation
Hi,
On Wed, Apr 20, 2011 at 2:01 PM, Brez Borland wrote:
> If I would ought to build the UAS, I would make it respond with 606 (Not
> Acceptable), and include Warning header with text description which
> parameter is missing. I would include the first missing parameter in the
> warning text. Furt
> In addition, if an SDP offer contains multiple streams (one RTP/AVP and
> one RTP/SAVP) those are actually *separate* streams, not alternate
> offers for the same stream.
>
> As far as I know, the only RFC-compliant way to offer both RTP/AVP and
> RTP/SAVP for the same media stream is through SDP
On Thu, Mar 31, 2011 at 5:21 PM, isshed wrote:
> yes Paul, you get the question right?
> do you think a client can send the 482...by client i mean a sip endpoint..
>
We do that in the python-sipsimple library in the following case: user
has added 2 accounts, and the library will use same local Co
> I would; I can't see a lot of value in rejecting an SDP offer because of a
> change like this. In Asterisk we've got a configuration option to not
> require SDP version number changes on received offers because there are at
> least two implementations out there that don't bother to increase the
>
Hi Kevin!
> I believe you are correct, the host address should not be changed in the
> 'o' line... it's not really used for anything except identifying the
> session anyway, there's no value in changing it.
>
I suspected that. :-)
> If the PBX in question is the one I think it is, this behavior
Hi,
While testing a SIP client I work on I noticed that there was some
interoperability issue with a certain PBX. This PBX will issue a
re-INVITE to both call legs once the call is established so that they
direct the RTP directly, without traversing the server. While doing
this, the origin SDP att
I'd do 305 in this one, but that's just my 2 cents.
On Thu, Mar 10, 2011 at 11:16 AM, isshed wrote:
> Hi All,
>
> If an initial INVITE from an endpoint offer contains the sdp as follows.
>
> m=audio 15190 RTP/AVP 100 101\r\n
> a=fmtp:18 annexb=yes\r\n
> a=fmtp:101 0-15\r\n
> a=rtpmap:100 UNACCEP
On Thu, Mar 3, 2011 at 2:11 PM, Yves Dorfsman wrote:
>
> rfc3261, section 10.2.3:
> A success response to any REGISTER request contains the complete list
> of existing bindings, regardless of whether the request contained a
> Contact header field. If no Contact header field is present in
On Tue, Mar 1, 2011 at 2:10 PM, mukesh vaidya wrote:
> I mean following steps i have done ..
>
> 1. Create 2 SIP gateways say A and B.
> 2. Set sdp-dir-attr-modif-mode to rfc3264 at gateway B.
What is that, a setting in the gateway?
> 3. Send an INVITE from A to B with SDP containing two directi
Yes and no :-) You can have a direction attribute on each media
stream, but you can't have multiple directions for a single stream,
what would that mean?
On Tue, Mar 1, 2011 at 12:47 PM, mukesh vaidya wrote:
> Can we pass multiple direction attributes in SDP in single pass ??
> _
On Tue, Feb 22, 2011 at 3:44 PM, Murad Ali wrote:
> Please unsubscribe my e-mail
>
> Thanks
> Murad
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Chec
> Does the ABNF for SIP uri's really support this?
>
AFAIS not:
hostport = host [ ":" port ]
host = hostname / IPv4address / IPv6reference
hostname = *( domainlabel "." ) toplabel [ "." ]
domainlabel = alphanum
/ alphanum *( alphanum / "-"
Hi,
On Thu, Feb 10, 2011 at 1:37 PM, wrote:
> Hi,
>
> I have this specific scenario and would like to know if its proper.
>
> We have a SIP Phone connected to our end which calls a third party SIP
> phone.
>
> The scenario is something like this:
>
> Our Phone 3rd Party
>
Hi,
> Agree. And to clarify a bit more:
>
> Important to note is that I think that you should never store the viewing
> version, like "olle@blåbärsmjölk.se" in infrastructure databases, nor handle
> it in the protocol messages. The software that shows URI's to users and
> accept user input will
On Wed, Feb 9, 2011 at 12:39 PM, Kevin P. Fleming wrote:
> The value of punycode is supposed to be that it is transparent to the
> proxy; it's just a text string that proxy will store, retrieve, compare,
> and eventually pass on to a DNS resolver. As long as the DNS resolver
> can handle transform
Hi all,
I've been working in unicode support with SIP and some doubts just
raised: lets say we have the following SIP URIs:
a) sip:me@saúl.com
b) sip:saúl...@example.com
For a) case should I encode the domain part using punycode, same as
it's done in email or web domains?
"saúl" gets translated
On Tue, Feb 8, 2011 at 5:05 PM, Iñaki Baz Castillo wrote:
> Hi, I've a problem with my DNS provider as it doesn't allow me to
> create a NAPTR as follows:
>
> aliax.net NAPTR 10 10 "s" "SIP+D2U" "" _sip._udp.aliax.net
>
> The problem occurs in the web panel when I fill the field
> "replacement
On Mon, Feb 7, 2011 at 1:04 PM, Nauman Sulaiman
wrote:
> Hi ,thamks. The thing is I have configured all Us and none of them have a
> proxy entry only domain. In the initial case here though the UA=> A simply
> sends it to b...@wherever.com as it doesn't know address of B.
>
Even if you don't
On Mon, Feb 7, 2011 at 12:42 AM, Nauman Sulaiman
wrote:
> Hi, situation is as follows
>
> 2 UA's A and B with private addresses communication with domain wherever.com
>
> A transfers B to Parking slot. Sometime later A retrieves B from parking
> slot. The Refer sent by proxy (at wherever.com) to
Hi,
On Thu, Feb 3, 2011 at 7:56 AM, prashant jain wrote:
> have a set up in which there are two sip client behind the
> NAT(restrictive) and through ICE they are establishing the connection.I have
> a douobt in the INVITE call establishment flow. Since SIP inherently have
> the capability tp trav
On Mon, Jan 31, 2011 at 6:36 PM, Mikko Lehto wrote:
> Worley, Dale R (Dale) wrote:
>
>> 2833-style DTMF is very common in "pure SIP" situations and is supported by
>> many telephone models.
>
> Sure.
> But what can you do when you have a requirement to 1) allow direct RTP
> and 2) to capture key
On Sat, Jan 22, 2011 at 8:19 PM, Chandan Kumar wrote:
>
> Hi,
>
> if we want to call a analog line on a PBX which does not act as a
> SIP-Server.Is there a way to do this?
> The call by IP INVITE look like "INVITE sip:192.168.10.234...".
> If we want to call a analog phone on a Peer to Peer P
> I thought of that - but what would we put there that everyone could support?
>
For the codec related 488, a 305 "Incompatible media format" could be
used. For IPv4/IPv6 stuff, 300 "Incompatible network protocol" or 301
"Incompatible network address formats" could help.
If none of them suit the
On Mon, Jan 10, 2011 at 6:08 PM, Olle E. Johansson wrote:
>
> 10 jan 2011 kl. 14.07 skrev Kevin P. Fleming:
>
>> On 01/10/2011 03:59 AM, Olle E. Johansson wrote:
>>> The draft changes the SDP offer/answer model so that an answer has to use
>>> the same protocol family (ipv4/ipv6) as the offer, wh
Hi,
On Fri, Dec 24, 2010 at 7:26 AM, kalpesh shukla wrote:
> hi,
>
> we are implementing MESSAGE method support on our IP PBX.
> i want to know how our PBX should respond to a MESSAGE request?
> Should i respond with final 200 OK or should i wait for the final response
> for the UA at the other
On Thu, Dec 16, 2010 at 3:39 PM, Chandan Kumar wrote:
> Hi ,
>
> . According to my understanding P-asserted Identity is added by Proxy servers
> within in the trusted domains . Please correct me if Iam wrong.
>
> one of our customers claims if P-asserted Identity is there in SIP header
> ,Incom
On Thu, Dec 16, 2010 at 3:58 PM, Wyne Wolf wrote:
> Yes. I did and we have a custom implementation. Our service facilitates
> client to client connection because traffic going through a server is
> absolutely unacceptable to our customers.
>
I'm curious, why did you go with a custom implementatio
On Thu, Dec 16, 2010 at 3:56 AM, Wyne Wolf wrote:
> I think ALG is useless. What we need is a protocol that the app can query
> the router about the public IP and Port of a mapping. Something like when
> the NAT entry is first created on the router, it sends back an ICMP packet
> containing the o
>> Can you provide a reference to this? Is it really allowed?
>
> Yes it is. Each direction of the stream can epecify its own payload
> types even for same codec. Also note that a stream could use codec A
> in left direction and codec B in right direction.
>
My question was not regarding the possi
> Also I believe that even static paylaod types can be used as dynamic ones.
> So lets say party A sends G711 a-law with payload type 0 but party B sends
> G711 a-law with dynamic payload 100 so what should happen in this case.
>
Can you provide a reference to this? Is it really allowed?
Regard
Hi,
On Mon, Nov 15, 2010 at 12:05 PM, Aaron Clauson wrote:
> -Original Message-
> From: sip-implementors-boun...@lists.cs.columbia.edu
> [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Saúl
> Ibarra Corretgé
> Sent: Monday, 15 November 2010 9:47 P
On Mon, Nov 15, 2010 at 11:35 AM, SIP Satan wrote:
> Which stateless proxy you are signaling with ? I thought it is very
> unlikely to have a forking capable stateless proxy. what is the need
> of such a stateless proxy( Only possible use case i see is load
> balancing :-) )?
>
I'm not testing/
On Mon, Nov 15, 2010 at 11:14 AM, SIP Satan wrote:
> A client not necessarily aware of what sort of proxy its talking
> to(stateless/state full). It can still follow the same rule
> irrespective of what kind of proxy it is talking to.
If this is so, this means that doing stateless forking would m
On Mon, Nov 15, 2010 at 10:56 AM, SIP Satan wrote:
> Proxy responsible for forking is supposed to filter the final
> responses. It only forwards the lowest final responses back to the
> user. Once a final responses is received caller can close down the
> transaction and ignore any other forked res
Hi,
After going through section 17.1.2 of RFC3261 and RFCs 4320 and 4321 I
still have some doubts about how a UAC should behave when it receives
a response to a NIT while on Completed state.
Lets assume the following: a UAC sent a SIP MESSAGE. Of course it
doesn't know what the proxy will do abou
Hi,
On Fri, Nov 12, 2010 at 12:17 PM, wrote:
> Hi All!
>
> I would like to consult about next scenario:
>
> UAC INVITE UAS
>
> UAC crashed (And we can do some finalization actions).
>
> What kind of finalization actions must/should UAC implement?
>
> In my implementation, if UAC craches, it
> In OMA-pres-rules, the order of "" elements is important as
> rules are checked from top to botton until one rule matches (its
> node).
>
According to OMA-TS-XDM-Core sec 6.6.2.3 first you need to compute all
the applicable rules and then decide which one will apply, so the
order doesn't really
> AFAIK there is a new condition introduced by OMA (OMA = IM/Presence
> for Mobile Operators obsessed with PushToTalk and private IP
> networks).
> That new condition is called "other-identity" [*] and its meaning is
> that any URI matches it.
>
Any URI which doesn't match any identity condition.
Hi Iñaki,
>
> Even if you get a good conclusion for this question, the problem would
> be: will your xcap server, your SIP presence server and all your SIP
> clients sharing the same account interpret such empty
> element in the same way? If not, you have a problem (as all the
> SIMPLE/XCAP implem
Hi,
Yesterday I came across the following doubt regarding pres-rules
documents and conditions:
According to the schema on RFC4745, the conditions element has a
min-coccurs of 0, so someone could add a rule without any condition at
all.
Now, section 10 says the following: "A rule matches if all c
Hi,
> I (my switch) am trying to detect this condition and, upon receipt of the
> ACK from Phone A to complete call setup, I (the switch) am attempting to
> issue 2 ReINVITEs (bumping the CSeq by 1), one to each Phone giving the
> others private LAN side IP / Port.
>
Do your endpoints and switch
Hi,
On Mon, Jun 14, 2010 at 12:13 PM, Iñaki Baz Castillo wrote:
> Hi, after reading RFC 3311 (UPDATE method) and RFC 4028 (Session
> Timers) it's not clear for me if I can use UPDATE requests and
> responses with no SDP offer/answers after the dialog is established.
>
> This is, does an UPDATE re
On Fri, Jun 4, 2010 at 5:42 AM, Bemali Wickramanayake
wrote:
> Hi,
> I have made a custom SDP application using RTC client, and we are testing it
> with Kapanga softphone.
> The problem we encountered is that it fails when it comes to SDP
> negotiations.
> The sample SDPs we have used to negotiate
Hi,
On Mon, May 10, 2010 at 5:16 PM, wrote:
> This is one of the primary motivating use cases behind SDP Capability
> Negotiation. See draft-ietf-mmusic-sdp-capability-negotiation-13, soon to
> be an RFC.
>
> Since it isn't a standard yet, the number of existing implementations will
> be few, s
Hi all,
While doing some reading today I got a bit confused about how to
express optional SRTP on an SDP. The right way seems to be to add 2
audio streams, one with SRTP and another one without:
- Stream without SRTP: RTP/AVP transport.
- Stream with SRTP: RTP/SAVP transport and a=cryplo line
On Tue, Apr 27, 2010 at 2:18 AM, Abu Ahmad wrote:
> HI,
> I would like to know which line from the below REGISTER packet have password
> value in MD5? Is it just plain password in MD5 or it comes with u...@pass?
> REGISTER sip:MySIPServer:6665 SIP/2.0Via: SIP/2.0/UDP
> 192.168.1.65;branch=z9hG4b
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