Re: [Sip-implementors] New Invite with same call id received when Non-Invite server is in completed state

Thank you Brett, Dale and Paul for your valuable comments. On Wed, Aug 24, 2016 at 9:05 PM, Paul Kyzivat wrote: > I agree with Brett on all points. If you are writing a test suite for sip, > then I think you should reject the invite. 481 seems like a reasonable > response, but may not be sufficie

Re: [Sip-implementors] New Invite with same call id received when Non-Invite server is in completed state

gt; On 24-Aug-2016, at 1:11 PM, isshed wrote: >> >>> On Wed, Aug 24, 2016 at 11:39 AM, My Gmail wrote: >>> This behavior is in correct. The device should match the dialog. If the >>> invite doesn't contain to tag then it is a new invite. If it's reinvite >

Re: [Sip-implementors] New Invite with same call id received when Non-Invite server is in completed state

1. > > > Thanks and Regards > Dheeraj Kumar > > Sent from iPhone > >> On 24-Aug-2016, at 11:19 AM, isshed wrote: >> >> Hi Folks, >> >> I am facing a strange problem. Below is the call flow. >> >> UA1-

[Sip-implementors] New Invite with same call id received when Non-Invite server is in completed state

Hi Folks, I am facing a strange problem. Below is the call flow. UA1---UA2 1) <= call is connected(callid1, ftag1, ttag1) > 2) <--

Re: [Sip-implementors] Offer answer model.... Updated

Thanks Paul. For details reply. Please find my response inline. On Wed, Jun 24, 2015 at 8:45 PM, Paul Kyzivat wrote: > On 6/23/15 10:23 PM, isshed wrote: >> >> Hi Paul, >> >> Below is the updated scenario. sorry for confusion by making step2 as >> recvonly. no

[Sip-implementors] Offer answer model

Hi Paul/Dale/Ankur, Thanks for your reply. below is the scenario please help resolve it. Below is the updated scenario. sorry for confusion by making step2 as recvonly. now it is fine. The problem here is that after step 9 call becomes audio only. Video disappears. UAC1--

Re: [Sip-implementors] Offer answer model

Thanks for response Dale. there is a typo fro 2nd message. read it as a=sendrecv. Meaning call get Successfully connected with audio and video. On Wed, Jun 24, 2015 at 8:44 AM, Dale R. Worley wrote: > isshed writes: >> UAC1---

[Sip-implementors] Offer answer model.... Updated

re that alters my reply. > > Thanks, > Paul > > > On 6/23/15 9:16 AM, isshed wrote: >> >> Hi All, >> >> Below is the scenario.. >> >> >> UAC1UAC2 >> >>

Re: [Sip-implementors] Offer answer model.... Updated

. What is happening is UAC2 is sending mline for video as last used(while holding) with valid port and a=sendonly?? Does RFC says anything or is it implementation dependent behavior?? Thanks, On Tue, Jun 23, 2015 at 6:43 PM, isshed wrote: > Hi All, &

[Sip-implementors] Offer answer model

Hi All, Below is the scenario.. UAC1UAC2 1)-INVITE (a=sendrecv)-> 2)<-200-OK(a=recvonly)- 3)---ACK--

Re: [Sip-implementors] SDP attribute for hold and rersume

;, then it > should offer "a=sendrecv" attribute, even if it had previously been forced > to answer something else. Without this behavior it is possible to get > "stuck on hold" in some cases, especially when a 3pcc is involved." > > >> -Original Me

[Sip-implementors] SDP attribute for hold and rersume

Hi All, Below is the scenario.. UAC1UAC2 1)-INVITE (a=sendrecv)-> 2)<-200-OK(a=recvpnly)- 3)---ACK--

Re: [Sip-implementors] Sessions Expires with Info

er...@corp.eastlink.caT: 519.786.1241 > > -Original Message- > From: sip-implementors-boun...@lists.cs.columbia.edu > [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Brett > Tate > Sent: March-27-15 9:21 AM > To: isshed; sip-implementors > Su

Re: [Sip-implementors] Sessions Expires with Info

1241 > > -Original Message- > From: sip-implementors-boun...@lists.cs.columbia.edu > [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of isshed > Sent: March-27-15 9:02 AM > To: Brett Tate > Cc: sip-implementors > Subject: Re: [Sip-implementors] Sessions Ex

Re: [Sip-implementors] Sessions Expires with Info

Thanks everyone for the responses. Brett, Lets assume the expires value is 100. the Update/Invite will go after 50 seconds after message (3). right? Thanks, On Fri, Mar 27, 2015 at 4:59 PM, Brett Tate wrote: >> The INFO can be out of dialog as well. > > INFO is a mid-dialog request. RFC 6086

[Sip-implementors] Sessions Expires with Info

Hi All, I have a scenario where my phone has negotiated with server as 60 seconds expires and UAC as refreshers. The Call flow with the server is as follows UAC-Server UAC-INVITE

Re: [Sip-implementors] SIP option tags are case sensitive?

Thank you Paul. On Tue, Dec 30, 2014 at 11:25 PM, Paul Kyzivat wrote: > On 12/30/14 7:06 AM, isshed wrote: >> >> Hi All, >> >> Could anybody please let me know if the SIP option tags are case >> sensitive? > > > No, they are not. > > RFC3261, se

Re: [Sip-implementors] SIP option tags are case sensitive?

t; [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of isshed > Sent: Tuesday, December 30, 2014 5:37 PM > To: sip-implementors > Subject: [Sip-implementors] SIP option tags are case sensitive? > > Hi All, > > Could anybody please let me know if the

[Sip-implementors] SIP option tags are case sensitive?

Hi All, Could anybody please let me know if the SIP option tags are case sensitive? Thanks, ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

Re: [Sip-implementors] Use case for P-Early-Media header (RFC 5009)

Thank you Brett for the reply. I am still not able to connect to any example. It would be great if you can provide an example. Thanks!! On Tue, Oct 7, 2014 at 4:24 PM, Brett Tate wrote: >> SIP defines the "P-Early-Media" header to control the forward/backward >> early media. Could any of you guy

[Sip-implementors] Use case for P-Early-Media header (RFC 5009)

Hi All, SIP defines the "P-Early-Media" header to control the forward/backward early media. Could any of you guys help me understand the use case? I am not able to get the use case. Why servers have policy to block/control the early media communication? Thanks in advance! Thank

Re: [Sip-implementors] Cancelling a request

Thanks Paul and Brett!! On Tue, Jul 15, 2014 at 8:34 PM, Paul Kyzivat wrote: > On 7/15/14 7:07 AM, Brett Tate wrote: >>> >>> Can there be any case when CANCEL reached to UA2 >>> before INVITE in case od UDP? because the 100 trying >>> can be sent by proxies as well. > > > Yes, 100 is sent hop by

Re: [Sip-implementors] Cancelling a request

Thanks Rahul and Tarun.. Can there be any case when CANCEL reached to UA2 before INVITE in case od UDP? because the 100 trying can be sent by proxies as well. On Tue, Jul 15, 2014 at 3:31 PM, Rahul Pathak wrote: > as per rfc 3261 you can send CANCLE message in this case.

[Sip-implementors] Cancelling a request

Hi All, I have a doubt in the following scenario. UA has sent INVITE to remote party. It received the 100 Trying. Now user wants to CANCEL the call. Can A CANCEL be sent at this point in time or it can not unless some non-100 provisional response comes? Thanks, Isshed

Re: [Sip-implementors] Dialog matching

considered? Thanks. On Wed, Apr 23, 2014 at 5:10 PM, isshed wrote: > Thanks Brett!! > > On Wed, Apr 23, 2014 at 4:26 PM, Brett Tate wrote: >>> Tags are all same i made a typo only the URI in >>> the ACK is changed. >> >> Hi, >> >> My response conce

Re: [Sip-implementors] Dialog matching

Thanks Brett!! On Wed, Apr 23, 2014 at 4:26 PM, Brett Tate wrote: >> Tags are all same i made a typo only the URI in >> the ACK is changed. > > Hi, > > My response concerning malformed messages still applies. The ACK is > malformed (from a RFC 3261 and RFC 4916 perspective); thus the device can

Re: [Sip-implementors] Dialog matching

Hey Brett, Tags are all same i made a typo only the URI in the ACK is changed. On Wed, Apr 23, 2014 at 3:47 PM, Brett Tate wrote: >> My phone has got an INVITE with the following field >> >> From: 1234 >> To: >> >> then it sends 200-INVITE as follows >> From: 123444 >> To: 432144 > > The 200 r

[Sip-implementors] Dialog matching

Hi All, My phone has got an INVITE with the following field From: 1234 To: then it sends 200-INVITE as follows From: 123444 To: 432144 Then my phone is receiving ACK as From: 123444 To: 432144 should my phone accept it and stop responding 200-INVITE? Thanks, ___

Re: [Sip-implementors] SDP offer answer model

I want to make it inter-operable with Polycom's real presence desktop. On Thu, Apr 17, 2014 at 6:43 PM, Paul Kyzivat wrote: > On 4/17/14 12:45 AM, isshed wrote: >> >> Thanks Paul and everyone ... >> >> I am designing Phone1 to use only one audio and one video. Pho

Re: [Sip-implementors] SDP offer answer model

ith > one audio and one video m-line, then post back with what you are trying to > accomplish, and we can discuss recommended ways of achieving that. > > Thanks, > Paul > > > On 4/16/14 2:51 AM, isshed wrote: >> >> Hi All, >> >> I have 1 b

[Sip-implementors] SDP offer answer model

Hi All, I have 1 basic query regarding ofer-answer model Phone1 is sending the offer with 2 audio mlines and 2 video mline like as follows. m=audio 3342 RTP/SAVP 0 8 127 a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:22 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/

[Sip-implementors] offer answer model

Hi All, Below is offer answer model between UA A and B. [Offer From A] v=0 o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com s= c=IN IP4 host.atlanta.example.com t=0 0 m=audio 49170 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PC

Re: [Sip-implementors] Query on reliable provisional response

ok thanks !! On Mon, Oct 28, 2013 at 5:03 PM, Brett Tate wrote: >> But since this is not truly a new request we >> have received, we are not sending an increased RSeq. >> I believe we are correct in this behaviour >> but how can we elicit a response from the SBC. > > I assume that you mean PRACK.

[Sip-implementors] Query on reliable provisional response

a new PRACK..? RFC 3262 also says that a UAC SHOULD NOT retransmit the PRACK request when it receives a retransmission of the provisional response" Also I want to tell you that our phone does not support RFC 3581. Is phone behavior justified? What should phone need to do here? Thanks, I

Re: [Sip-implementors] SDP Offer Answer Model

Thanks Paul and Brett. I have treated as a valid offer because the port is zero. Also, if all of the m lines presents do not have PT list. then I rejected with 488. Thanking you again!! On Wed, Oct 23, 2013 at 8:03 PM, Paul Kyzivat wrote: > On 10/23/13 6:23 AM, Brett Tate wrote: > >> Also I w

Re: [Sip-implementors] SDP Offer Answer Model

Also I want to know what should be the answer in this case ? On Wed, Oct 23, 2013 at 10:21 AM, isshed wrote: > Thanks Brett!! > > > On Tue, Oct 22, 2013 at 10:47 PM, Brett Tate wrote: > >> > Can there be an offer with as follows. >> > >> >

Re: [Sip-implementors] SDP Offer Answer Model

Thanks Brett!! On Tue, Oct 22, 2013 at 10:47 PM, Brett Tate wrote: > > Can there be an offer with as follows. > > > > v=0 > > o=abc 940493389 1 IN IP4 10.10.10.10 > > s=- > > c=IN IP4 10.10.10.10 > > t=0 0 > > m=audio 0 RTP/AVP > > > > Here m line does not ha

[Sip-implementors] SDP Offer Answer Model

Hi all, Can there be an offer with as follows. v=0 o=abc 940493389 1 IN IP4 10.10.10.10 s=- c=IN IP4 10.10.10.10 t=0 0 m=audio 0 RTP/AVP Here m line does not have any payload format . Is this a valid offer? what is the use case of this offer? Thanks,

Re: [Sip-implementors] Forking scenario handling at UA Endpoint

p; regards > Ankur Bansal > > > On Tue, Sep 10, 2013 at 12:29 PM, isshed wrote: > >> *Hi All,* >> * >> * >> *Below is the scenario we need to implement for our client. This is a >> forking scenario.* >> * >> >> UA

[Sip-implementors] Forking scenario handling at UA Endpoint

*Hi All,* * * *Below is the scenario we need to implement for our client. This is a forking scenario.* * UA Proxy **| INVITE | **|————>| **| 100 Trying | **|<| **| 180 Ringing: To tag=A |

Re: [Sip-implementors] Query for "received" parameter in top Via header

Anand the received parameter is added by your own stack(one who received the request). It contains real IP from where the packet is received. In case of NAT it will be NAT IP. On Tue, May 21, 2013 at 2:35 PM, ANAND KUMAR wrote: > Section 18.2.2 Sending Responses (For servers) of rfc3261 says: >

Re: [Sip-implementors] Certification Agency to test my SIP Stack

Thanks praveen, for the info. We can provide our device (it's a sip client product device )..We are ready to pay for the testing. On Fri, Feb 22, 2013 at 12:40 AM, Praveena Ss wrote: > hi isshed, > > i don't think any labs/organizations do only sip stack testing...but you &g

Re: [Sip-implementors] Certification Agency to test my SIP Stack

aveena Ss wrote: > >> hi isshed, >> >> i don't think any labs/organizations do only sip stack testing...but you >> can do testing with so many open source available sip clients and servers >> [open ims]. >> >> usually labs/organizations do sip product(ei

Re: [Sip-implementors] Certification Agency to test my SIP Stack

I have come to know about University of New Hampshire upon browsing on net. UNH-IOL does interoperability between different vendors(UA, Proxy, AS). Are there anyother Labs/organization/companies which do the same? Thanks. On Wed, Feb 20, 2013 at 7:52 PM, isshed wrote: > Hi All, >

[Sip-implementors] Certification Agency to test my SIP Stack

Hi All, Could anybody please suggest me the names of some Certification Agencies which can test my sip stack? I have implimented an IMS based SIP client. I want it to be mature enough to compete the SIP STACK present in the market. Any suggestions would be appreciated. Thanks, Isshed

[Sip-implementors] Certification Agency to test my SIP Stack.

Hi All, Could anybody please suggest me the names of some Certification Agencies which can test my sip stack? I have implimented an IMS based SIP client. I want it to be mature enough to compete the SIP STACK present in the market. Any suggestions would be appreciated. Thanks, Isshed

Re: [Sip-implementors] 416 Unsupported Uri Scheme

Thank you Brett!! On Fri, Feb 1, 2013 at 1:25 AM, Brett Tate wrote: > > If suppose first INVITE message and To header are of > > tel URI type. And the server responds with 416. Should > > we not be changing the To header from tel URI to > > SIP URI in this case? > > According to RFC 3261, the r

[Sip-implementors] 416 Unsupported Uri Scheme

3261. Please help me understand this scenario. Is there any other specs/rfc for translating tel URI to sip sip URI other than RFC 3261/3966? Thanks, Isshed. ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.

Re: [Sip-implementors] Processing Update without SDP when an Update is pending

Hi Deepak, If the UPDATE contains neither SDP nor session refress parameter(timer). I guess you can respond with 200 OK. What is the harm in replying with 200 Ok in this case? Thanks. On Wed, Jan 16, 2013 at 6:39 PM, Brett Tate wrote: > RFC 3311 section 5.2: > > A UAS that receives an UPDATE

Re: [Sip-implementors] SIP Testing

ately check for dialing without proxy using sipp without any > config. > > > Regards, > > Vineet Menon > > > > > On 25 January 2012 10:38, isshed wrote: > >> Hello All, >> >> We have implemented a SIP user agent which supports dual stack(it can >

[Sip-implementors] SIP Testing

Hello All, We have implemented a SIP user agent which supports dual stack(it can support IPv6 and IPv4 both at the same time). I want to test it.. like signalling on IPv4 and media is travelling on IPv6. Is there any tool available? Thanks, ___ Sip-impl

[Sip-implementors] IPsec

Hi All, Sorry to bother you but I am sure many of you are implementing the IPsec. I am very new to this technology. Today I was writing a simple program of key management using PF_KEY sockets.Basically I downloaded unixnetworkprograming books example from internet. while compiling I am getting the

Re: [Sip-implementors] AOR matching

Thanks Guys for all your valuable comments! On Fri, Oct 7, 2011 at 2:05 AM, Paul Kyzivat wrote: > At end > > On 9/23/11 11:35 AM, Worley, Dale R (Dale) wrote: > >> From: isshed [isshed@gmail.com] > >> > >> I have registered my UAC with AOR and Cont

Re: [Sip-implementors] AOR matching

iginal Message- > From: sip-implementors-boun...@lists.cs.columbia.edu [mailto: > sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext isshed > Sent: Friday, September 23, 2011 10:53 AM > To: sip-implementors > Subject: [Sip-implementors] AOR matching > > Hi All,

[Sip-implementors] AOR matching

. My UAC returns 403 forbidden since it does not match the AOR. Is it a valid behavior? can you please provide the reference for the same? Thanks, Isshed ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbi

[Sip-implementors] Subscribe transaction

Hi All, Can any one tell me the behavior of the scenario as below? UAC===UAS 1. ===SUBSCRIBE== 2. <200-SUBSCRIBE= 3.

[Sip-implementors] SDP negotiation

103 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:99 RED/8000 a=rtpmap:104 PCMA/8000 a=rtpmap:103 t38/8000 a=fmtp:103 T38FaxVersion=0;T38MaxBitrate=14400;T38FaxMaxDatagram=173;T38FaxRateManagement=transferredTCF a=fmtp:99 104/104 a=fmtp:99 103/103 Thanks, Isshed

[Sip-implementors] Call Transfer In dialog REFER

###<---BYE/200-> My confusions are. 1. Do REFER message contains Require: tdialog? Is it mandatory to transfree support tdialog? 2. Do REFER message contains Target-Dialog (i.e. dialog 1)..I think it is not required in In-dialog refer. 3. How and

Re: [Sip-implementors] Call Transfer Using REFER

Thanks Guys for providing me this valuable information. Currently I am not supporting Referred-By header instead Target-Dialog is being supported. does this make any change in the message? Obviously there is this header present. Thanks. On Wed, Apr 27, 2011 at 5:12 AM, Iñaki Baz Castillo wrote:

Re: [Sip-implementors] Call Transfer Using REFER

...@lists.cs.columbia.edu > [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext > isshed > Sent: Tuesday, April 26, 2011 12:10 PM > To: Worley, Dale R (Dale) > Cc: sip-implementors > Subject: Re: [Sip-implementors] Call Transfer Using REFER > > Thanks Dale for you

Re: [Sip-implementors] Call Transfer Using REFER

Thanks Dale for your response. so the other doubt is what will happed to this dialog when the call gets transferred. does it not get destroyed with the BYE? if so what about the rest of the notify? I know I am asking the basics but it will improve my understanding. Thanks. On Tue, Apr 26, 2011 a

[Sip-implementors] Call Transfer Using REFER

Hello All, I want to implement call transfer feature on a user agent. As you all know it can be done in 2 way. 1. sending REFER out of dialog and 2. sending REFER in dialog. As rfc 3515 says " A REFER request MAY be placed outside the scope of a dialog created with an INVITE. " Also as per RFC 3

Re: [Sip-implementors] 482 loop detected.

yes Paul, you get the question right? do you think a client can send the 482...by client i mean a sip endpoint.. Thanks On Thu, Mar 31, 2011 at 5:29 PM, Paul Kyzivat wrote: > Is this a trick question? > > A *client* never sends responses. The thing that sends (any) response is > an server. > > I

[Sip-implementors] 482 loop detected.

Hi All, Is there any scenario or use case where a sip client can send 482. Thanks, ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

[Sip-implementors] UAC BYE handling

Hi ALL, There is a dialog established between UAC and UAS. Now UAC has send BYE to terimate the dialog. Before getting the BYE from network the UAS sends the BYE to UAC as well. Now what should be the behaviour of UAC. should it response a 2xx response or 481 or some other respose. UAC <

[Sip-implementors] sip server

Hi All, I have implemented a sip client. I want tot test it(negative scenario) with some freely available server. Is there any good server. I would like to modify the message in the server. Can anybody suggest one good sip server. Thanks. ___ Sip-implem

Re: [Sip-implementors] + prefix in To header.

? Thanks, On Mon, Mar 14, 2011 at 5:02 PM, Kevin P. Fleming wrote: > On 03/14/2011 12:41 AM, isshed wrote: > > Hello All, > > > > What should be the behavior of server in the following case. > > > > 1. Client has registered with url as +16035551...@open-ims.test

Re: [Sip-implementors] + prefix in To header.

On Mon, Mar 14, 2011 at 11:11 AM, isshed wrote: > Hello All, > > What should be the behavior of server in the following case. > > 1. Client has registered with url as +16035551...@open-ims.test. > > now client is trying to make a call. > > 1. Request uri of Invite was having

[Sip-implementors] + prefix in To header.

Hello All, What should be the behavior of server in the following case. 1. Client has registered with url as +16035551...@open-ims.test. now client is trying to make a call. 1. Request uri of Invite was having +*16035551...@open-ims.test* but 2. To url of Invite was having *16035551...@open-ims

[Sip-implementors] SIP message size

Hi All, What is the maximum message size that can go on UDP. Why is rfc 3261 mandates that a large size message should be sent on TCP? Thanks. ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslis

Re: [Sip-implementors] [Sip] Warning header

Thank for your response. As you correctly said the warning code 306 indicates that the recipient cannot understand one of the a= lines. But in this case recipient is understanding the a=rtpmap:100 UNACCEPTABLECODEC/8000 line. Only thing it is not understanding is the media format(i.e. encoding typ

Re: [Sip-implementors] [Sip] Warning header

out the response. > > Regards, > > Christer > > > > > > > > >From: sip-boun...@ietf.org [mailto:sip-boun...@ietf.org] On Behalf > Of isshed >Sent: 10. maaliskuuta 2011 13:17 >To: s...@ietf.org; sip-implementors >

[Sip-implementors] Warning header

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 UNACCEPTABLECODEC/8000\r\n a=sendrecv the terminating endpoint returns an error response 488 with a warning header as follo

Re: [Sip-implementors] Content-Disposition

If this header in initial invite and parsing fails. does the call get connected? On Wed, Mar 9, 2011 at 11:16 PM, isshed wrote: > does anyone know what the behaviour if parsing of Content-Disposition > header fails. > > Thanks, > HArendra > _

[Sip-implementors] Content-Disposition

does anyone know what the behaviour if parsing of Content-Disposition header fails. Thanks, HArendra ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

Re: [Sip-implementors] Different SDP Session Version in 183 & 200 OK - Email found in subject

hello nitin,, you can not increase session version. session version can only be incremented. as per RFC 4566 " is a version number for this session description. Its usage is up to the creating tool, so long as is increased when a modification is made to the session data. Again, it is RECOMMENDED

[Sip-implementors] Sending Register to an UA

What error code should be returned by a UA if we send a REGISTER. I am implementing a sip client. Thanks ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors