Re: [Sip-implementors] Regarding Subscription Termination

Ok, I got it. Thanks for your support. Thanks and Regards, Amith -Original Message- From: Brett Tate [mailto:br...@broadsoft.com] Sent: Tuesday, July 05, 2011 5:05 PM To: Amith R R; Sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Regarding Subscription Termination

Re: [Sip-implementors] Regarding Subscription Termination

, 2011 4:13 PM To: Amith R R; Sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Regarding Subscription Termination > But, in RFC 5589, it is specified that we can terminate > the subscription using BYE. The above statement is not a correct interpretation of the RFC 5589 s

Re: [Sip-implementors] Regarding Subscription Termination

-Original Message- From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Amith R R Sent: Monday, July 04, 2011 9:19 AM To: Sip-implementors@lists.cs.columbia.edu Subject: [Sip-implementors] Regarding Subscription Termination using B

[Sip-implementors] Regarding Subscription Termination using BYE

Hi In RFC 5057, it is mentioned that BYE cannot be used to terminate the subscription created by REFER. Normal termination of a usage has no effect on other usages sharing the same dialog. For instance, terminating a subscription with a NOTIFY/Subscription-State: terminated will

Re: [Sip-implementors] Multiple early media sessions within a same dialog

Is it possible in real time to establish two early session (with same tag) > between UAC and the soft switch ? > > > > Regards, > > Harlin > -- > > *From:* $...@r\/|>r!`/@ [mailto:sarvpriyagu...@gmail.com] > *Sent:* Tuesday, January 0

Re: [Sip-implementors] Multiple early media sessions within a same dialog

Hello guys, I gues he is only referring multiple announcements within the same dialog. He is not referring to multiple UACs. Harlin, Will you please explain iwhat exactly is your scenario. Is it different announcements to different UACs or multiple announcements to one UAC? cheers!! sarvpriya O

Re: [Sip-implementors] dns record updation

Hi, No. You need to re-query for the record which has expired. All other records hold valid. cheers!! sarvpriya On Tue, Dec 21, 2010 at 3:23 PM, Satyakumar wrote: > Hi, >Do we need to do a fresh DNS record (SRV/A)query, when ttl of one > record lapsed, among a group of cached records. >

Re: [Sip-implementors] DNS Round Robin

the second server. SG-> the above condition of DNS records expiry takes a priority and you need to send the request as per the new results. cheers!! sarvpriya On Thu, Nov 11, 2010 at 5:19 PM, Alex Hermann wrote: > On Thursday 11 November 2010, $...@r\/|>r!`/@ wrote: > > Hi,

Re: [Sip-implementors] DNS Round Robin

Hi, If you are sending second invite with Authorization fields then you should send it to the first one only. Also the records are returned in priority. Thus you should first try on the first server only. Are you able to provide authorization info. If no then you can send a new invite to second s

Re: [Sip-implementors] Contact: * - in 200 OK of REGISTER

Hi, why does the registrar wants to put * in the response? The contact uri in 200 respose is to tell its contact uri to the UAC. I guess the sip grammar will not allow it. Any particular reason you want to do this?? cheers!! sarvpriya On Tue, Oct 12, 2010 at 4:28 PM, hanifa.mohammed < hanifa.moh

Re: [Sip-implementors] UAS behavior : Multiple 18x messages

1. Can UAS responds with multiple 18x for the INVITE request received? YES. 2. What should be the behavior of UA1 if it receives multiple 18x? Create dialog if the response contains tag. Update your state machine. 3. What should be the behavior of UA1 while receiving the second 18x with SDP? When t

Re: [Sip-implementors] Via or Route header to send response to the requestor at Proxy

velled through and on UAS used to identify the nodes the > response should be travelling through. > 2. Record header is build based on record-route (which every proxy adds in > the Initial request) and used for future request/responses. > > Is this the correct understanding? > >

Re: [Sip-implementors] Via or Route header to send response to the requestor at Proxy

Hi Santosh, The responses are always send as per VIA only. When the initial request as in which can establish a dialogue is coming from UAC to UAS via a proxy, during the traversal, each hop's IPaddress is appended in the VIA header field. The hop's are can be determined by pre existing route set

Re: [Sip-implementors] Detection of Cable disconnections in SIP

I think this is going bit offtopici think the forum is not meant to discuss my best bets. On Thu, Oct 7, 2010 at 10:45 AM, Alex Balashov wrote: > It really sounds like session timers (RFC 4028) are your best bet. > > On 10/07/2010 12:33 AM, $...@r\/|>r!`/@ wrote: > > >

Re: [Sip-implementors] Detection of Cable disconnections in SIP

l Message- > From: sip-implementors-boun...@lists.cs.columbia.edu [mailto: > sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext > $...@r\/|>r!`/@ > Sent: Thursday, October 07, 2010 10:03 AM > To: Worley, Dale R (Dale) > Cc: Manoj Priyankara [TG]; sip-implementors@lists.c

Re: [Sip-implementors] Detection of Cable disconnections in SIP

Hi, Its not always a voice callin video calls we would need such mechanism. cheers!! sarvpriya On Wed, Oct 6, 2010 at 7:27 PM, Worley, Dale R (Dale) wrote: > > From: sip-implementors-boun...@lists.cs.columbia.edu [ > sip-implementors-boun...@lists.cs

Re: [Sip-implementors] Detection of Cable disconnections in SIP

To add to Alex answer, RFC 4028 suggest a solution by which if your UA supports the RFC then timeouts can be managed. Thus even if another NE do not support it and your UA does, you will be able to detect any broken connection. cheers!! sarvpriya On Wed, Oct 6, 2010 at 11:40 AM, Alex Balashov wro

Re: [Sip-implementors] Open Source SIP stack

Hi, You can also use OSIP. Its in C plus it exposes pointers to function hence helps in customisation. Here is the link http://www.gnu.org/software/osip/ There is an OSIP implementors forum as well. cheers!! sarvpriya On Tue, Oct 5, 2010 at 10:51 PM, Bossiel t

Re: [Sip-implementors] sip to isup interworking(Q.1912.5)

hi, SIPP has support for SIP-I as well. cheers!! sarvpriya On Wed, Sep 15, 2010 at 12:11 PM, Parveen Kumar Jain wrote: > Hi Honsha, > Thanks for your quick reply. > > As far as I know , sip-p is used only for testing the sip > applications(using > some scripts).Can you please elaborate a bit

Re: [Sip-implementors] Query on Forking

Hi ALL, till you dont receive the tag in 180, you are not creating any dialogs. Thus when you receive 180 WO to tag you will not create any dialog and just update the call state . The concept of forking and stuff comes when you receive multiple 18x with different To-tags. cheers!! sarvpriya On T

Re: [Sip-implementors] Invite without SDP

Hi, its always recommended to send the SDP to hold a call. There are two ways of acheiving this 1) Changing the port to zero for the media lines 2) Or marking the stream as sendonly. cheers!! sarvpriya On Thu, Aug 19, 2010 at 12:12 PM, Manoj Priyankara [TG] wrote: > Dear All, > > If a particula

Re: [Sip-implementors] Session Timer Query

e minimum timer for > the server. The 422 response MUST contain a Min-SE header field with the > minimum timer for that server. > > > On Thu, Aug 19, 2010 at 12:21 PM, $...@r\/|>r!`/@ > wrote: > >> Hi, >> >> The way session timers work is >> 1)

Re: [Sip-implementors] Session Timer Query

Hi, The way session timers work is 1) if CHASIS2 doesnt accept this values and want to decrease them, it needs to send 422 response with its supported values. Then its upto CHASIS1 to decide whether it accepts the same or not. 2) Any UA can only increase the value of MinSE as its the minimum bound

Re: [Sip-implementors] Transport Switching scenario by proxy

Thanks a lot :-) On Mon, Aug 16, 2010 at 9:19 PM, marius zbihlei wrote: > Paul Kyzivat wrote: > >> $...@r\/|>r!`/@ wrote: >> >> >>> Hi All, >>> >>> Can you please refer me a write up which explains how a proxy handles >>>

[Sip-implementors] Transport Switching scenario by proxy

Hi All, Can you please refer me a write up which explains how a proxy handles transport switching scenarios if required which frwding request. Sent from BlackBerry® on Airtel ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu http

Re: [Sip-implementors] SIP OPTIONS "ping"

Hi, Gud to see your initiative. Instead of using 486 for denoting overload, 503 message can be used. I have seen this usage to use as overload control mechanism. Entirely in my opinion, 486 is not apt. cheers sarvpriya http://sarvpriyak.blogspot.com/ On Thu, Aug 12, 2010 at 10:44 AM, Paul

Re: [Sip-implementors] SDP in re-invite

Hi All, You should really check whether its a new invite or Reinvite. With the same dialog credentials, its should not be the case of new dialog. Also if UA sends you a reinvite without SDP, then you have to send the offer in 2xx to Invite. Then which SDP you will send...already negotiated one or

Re: [Sip-implementors] Record-Route question

Hi, Infact the ACk message would also go without Record-Route. When the UAC receives 2xx for Invite, it initialises all the dialog parameters like conversion of record route to route, new contact uri. cheers!! sarvpriya On Thu, Aug 12, 2010 at 1:43 AM, fuliang yuan wrote: > Hi, > > Here is the

Re: [Sip-implementors] Re-registration and Registration renewal

Hi, it should be with the same call-id. The rfc clearly (section 10)specifies that the call-id should be same for all registrations during a single reboot. cheers!! sarvpriya *http://sarvpriyak.blogspot.com/ * On Thu, Jul 29, 2010 at 3:07 AM, Vivek Singla wrote: > I did. But it doesn't specify

Re: [Sip-implementors] Request URI

Hi Nitin, The % symbol is allowed in the SIp grammar. below is an example uri straight from the RFC sips:al...@atlanta.com?subject=project%20x&priority=urgent Now ruling this out as the problem area...the second invite is not a reinvitethere is a diff b/w reinvite and new invite txn. Kndly c

Re: [Sip-implementors] RFC 3261, 100 Trying on Re-Invite

Hi, This problem occurs not only in ReInvite but in INVITE transactions also. Its a bug in SIP and has been reported as well. In our implementation, we didnt hamper the existing timers but created a new one whose value we set > timerB. For sure the dialog needs to be terminated. cheers!! sarvpriy

Re: [Sip-implementors] 302 direct by Proxy violating spec??

Hi, Can you please elaborate the call flow after CANCEL has been sent. cheers!! sarvpriya http://sarvpriyak.blogspot.com/ On Tue, Jul 27, 2010 at 2:53 AM, Nauman Sulaiman wrote: > Hi , we have a set up where we are using a SIP proxy( OpenSIPS) to handle a > 302 redirect locally but it doesn't

Re: [Sip-implementors] Switching frm UDP to TCP while sending response

NO, its not valid for responses. For sending responses, same protocol must be used. On Fri, Jul 23, 2010 at 4:40 PM, hanifa.mohammed < hanifa.moham...@globaledgesoft.com> wrote: > Hi all, > > Is the below requirement frm RFC3261 applicable for response also? If yes, > and if the VIA > of the res

Re: [Sip-implementors] retransmission of 180Ringing with require: 100rel

Just check the tag in the 180's. The 180s might be of sum other UAS(typical forking scenario). Also if you have received 2xx then you can very well ignore 1xx msg. On Thu, Jul 22, 2010 at 1:07 AM, WORLEY, Dale R (Dale) wrote: > > From: sip-implementors-bou

Re: [Sip-implementors] retransmission of 180Ringing with require: 100rel

the UAS cannot send CANCEL request. The UAC can only send CANCEL to terminate. After receiving 500 final response from UAS, UACsends ACK and after the timer expiry, Invite transaction will be terminated. On Wed, Jul 21, 2010 at 3:13 PM, Md Faruk Apel Chowdhury < mdchowdh...@etisalat.ae> wrote: >

Re: [Sip-implementors] retransmission of 180Ringing with require: 100rel

yes yes in that case it shud send PRACK. I am sorry in the first mail it was the same rseq. Apologies for ignoring. On Wed, Jul 21, 2010 at 2:57 PM, Iñaki Baz Castillo wrote: > 2010/7/21 $...@r\/|>r!`/@ : > > > For UAC behavior section 4 says > > " Handling of subse

Re: [Sip-implementors] retransmission of 180Ringing with require: 100rel

Hi Peter, RFC 3262 for UAS section 3 clearly says "After the first reliable provisional response for a request has been acknowledged, the UAS MAY send additional reliable provisional responses. The UAS MUST NOT send a second reliable provisional response until the first is acknowledged. After the

Re: [Sip-implementors] 408 or CANCEL

Hi, if its ringing then 18x has already been received. Please correct me if I am wrong. On Fri, Jul 9, 2010 at 10:40 AM, Harbhanu wrote: > >>CANCEL to the originator? Of course not. The proxy would send a 408 or > >>480 to the calling UA (to the UAC) and CANCEL to all the ringing > >>UAS's. > >

Re: [Sip-implementors] Retrans 2xx handilng - RFC-4028

Hi Harbhanu, For your scenario, you should just send ACK to the retransmitted 2xx. If u keep on rcving the retx 2xx, send ack till the ACK timer expires. As far as clubbing session timer with retx 2xx is wrong. Session refresh request should be sent as per agreed time. Also u cannot reset/rehandle