Re: [Sip-implementors] Sip-implementors Digest, Vol 98, Issue 11

> > > Daftar: https://bit.ly/PertaminaGroup > ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

Re: [Sip-implementors] Sip-implementors Digest, Vol 98, Issue 9

DAFTAR: https://bit.ly/bumnPertamina > > > ___ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors _______

Re: [Sip-implementors] Sip-implementors Digest, Vol 98, Issue 9

> > DAFTAR: https://bit.ly/bumnPertamina > ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

Re: [Sip-implementors] Sip From header without user part

> > > > As per ABNF grammar, the above one is also should not be an issue > > as From > > > header but could let me know if I misinterpreted the ABNF grammar? > > > > > > BR///Rakesh > > > _

Re: [Sip-implementors] Sip From header without user part

Rakesh writes: > I could see in a sip CANCEL message From Header as below > > From: "test" ;tag=3bbb9483d215d830c635372f8f181929 > > is this correct? To summarize what Paul has said: Syntactically, this header is valid. But usually the question is whether the entire SIP message containing the he

Re: [Sip-implementors] Sip From header without user part

mentors@lists.cs.columbia.edu <mailto:Sip-implementors@lists.cs.columbia.edu> > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors <https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors> ___ Sip-implemento

Re: [Sip-implementors] Sip From header without user part

s also should not be an issue as From > > header but could let me know if I misinterpreted the ABNF grammar? > > > > BR///Rakesh > > _______ > > Sip-implementors mailing list > > Sip-implementors@lists.cs.columbia.edu > &

Re: [Sip-implementors] Sip From header without user part

__ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors _______ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/li

[Sip-implementors] Sip From header without user part

Hi Team, I could see in a sip CANCEL message From Header as below From: "test" ;tag=3bbb9483d215d830c635372f8f181929 is this correct? Normally the From header is sip:user@domain As per ABNF grammar, the above one is also should not be an issue as From header but could let me know if I misinter

Re: [Sip-implementors] SIP over QUIC

iences on that topic, or any pointers/hints on non-trivial things that > we should take extra care of. > > /Filippos > ___ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.colum

[Sip-implementors] SIP over QUIC

Hello everyone, Has anyone implemented SIP over QUIC, or started looking at that ? We will start looking at that soon, so I was wondering about other people's experiences on that topic, or any pointers/hints on non-trivial things that we should take extra care of. /Filippos __

Re: [Sip-implementors] SIP REGISTER MESSAGE flow

Hi Ankur, Sure will review the RFC Thanks for you reply On Tue, 30 Nov 2021 at 11:00 AM, ankur bansal wrote: > @arun.taga...@gmail.com > Please check RFC 5626 3.3 > . Multiple > Connections from a User Agent > I believe you might find

Re: [Sip-implementors] SIP REGISTER MESSAGE flow

ase signalling load on NW, but want to understand > is > >> this flow allowed as per rules ? Or there are any restrictions please > help > >> on understanding > >> > >> > >> > >> -- > >> > >> With Regards > >> > >> Aru

Re: [Sip-implementors] SIP REGISTER MESSAGE flow

Thanks Ranjit & Paul, Yes, NW can send ‘n’ number of PCSCF addresses, Yes, agree Ranjit it’s may be not practical but just wanted to understand if that is allowed? Or Nw accepts REG in sequence like Reg to 1 PCSCF Fail/error response from NW Then retry on 2PCSCF address Fail/error response from

Re: [Sip-implementors] SIP REGISTER MESSAGE flow

gt; +91 9449 029729 > _______ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

Re: [Sip-implementors] SIP REGISTER MESSAGE flow

Arun, Your query doesn't have enough context to understand the case. It does however appear to more likely to be something that should be answered by someone in the 3GPP IMS community. If you want an answer here please spell out your case in much more detail. Thanks, Paul On

[Sip-implementors] SIP REGISTER MESSAGE flow

Hi Team, Need some info, can UE be allowed (as per rule) to send SIP REGISTER requests to all the received P-CSCF addresses from NW (during attach) ? For Ex. If UE received 5 P-CSCF address during attach Is it allowed to send REG to address at the same time UE REG to P-CSCF-1---> UE REG to P-CSC

[Sip-implementors] SIP Forum Announces SIPit 33 - Focused on STIR/SHAKEN Interop Testing

The SIP Forum is proud to announce that our next SIP Interoperability Testing Event, SIPit 33, has been scheduled the week of December 13-17, 2021. Organized by the SIP Forum, the SIPit events are the world's premier interoperability testing events for SIP, bringing together leading SIP applic

Re: [Sip-implementors] Sip-implementors Digest, Vol 85, Issue 3

STOP -Original Message- From: Sent: Thu, 03 Jun 2021 12:00:01 -0400 To: 7076373...@txt.att.net Subject: Sip-implementors Digest, Vol 85, Issue 3 >Send Sip-implementors mailing list submissions to > sip-implementors@lists.cs.columbia.edu > >To subscribe or unsubscribe via t

Re: [Sip-implementors] Sip-implementors Digest, Vol 85, Issue 2

STOP -Original Message- From: Sent: Wed, 02 Jun 2021 12:00:01 -0400 To: 7076373...@txt.att.net Subject: Sip-implementors Digest, Vol 85, Issue 2 >Send Sip-implementors mailing list submissions to > sip-implementors@lists.cs.columbia.edu > >To subscribe or unsubscribe via t

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 4

_ > > Sip-implementors mailing list > > Sip-implementors@lists.cs.columbia.edu > > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > > -- > Jonathan Lennox > len...@cs.columbia.edu > sip-implementors-ow...@cs.columbia.edu > ___ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 4

I am the list administrator, and I've removed this user. Sorry about the inconvenience! On Friday, January 8 2021, "Alex Balashov" wrote to "sip-implementors@lists.cs.columbia.edu" saying: > I am not a list administrator, but speaking as a civilian: > > I think we have been patient, but the ti

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 4

ov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > _______ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 4

I am not a list administrator, but speaking as a civilian: I think we have been patient, but the time is nigh to mute this subscriber. On 1/8/21 1:12 AM, supp...@advisory24.vip wrote: Hi Sip-implementors,  May I have your order number please? And please offer the email address and full na

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 4

Hi Sip-implementors,  May I have your order number please? And please offer the email address and full name that you filled in when you made this order on our website. So I can check it for you.  Cathy Lin On

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 4

Thank you for contacting us! We have received the email. We are really sorry that we're off work now. Our working time is from 10am to 7pm (GMT 9), Mon to Fri. Please do not worry. We will arrange a dedicated customer service to solve your problem within 12 hours. We will try our best to he

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 3

Thank you for contacting us！ May I have your order number please? And please offer the email address and full name that you filled in when you made this order on our website. So I can check it for you. ___ Sip-implementors mailing list Sip-impleme

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 2

Thank you for contacting us! We have received the email. We are really sorry that we're off work now. Our working time is from 10am to 7pm (GMT 9), Mon to Fri. Please do not worry. We will arrange a dedicated customer service to solve your problem within 12 hours. We will try our best to he

Re: [Sip-implementors] Sip-implementors Digest, Vol 81, Issue 1

Thank you for contacting us！ May I have your order number please? And please offer the email address and full name that you filled in when you made this order on our website. So I can check it for you. ___ Sip-implementors mailing list Sip-impleme

Re: [Sip-implementors] Sip-implementors Digest, Vol 79, Issue 1

Thank you for contacting us! We have received the email. We are really sorry that we're off work now. Our working time is from 10am to 7pm (GMT 9), Mon to Fri. Please do not worry. We will arrange a dedicated customer service to solve your problem within 12 hours. We will try our best to he

[Sip-implementors] SIP TLS client cert

Hi! I hope everyone stays safe in these times. During this week’s IETF I ended up in a discussion about using TLS client certs in SIP. I have been testing this a long time ago, but obviously not fully. The question I got I failed to find an answer to, which is annoying :-) Here it goes, let’s s

Re: [Sip-implementors] Sip-implementors Digest, Vol 77, Issue 2

Hi Sip-implementors, Thank you for contacting us！  May I have your order number please?  And  please offer the email address and full name that you filled in when you made  this order on our website.  So I can check it for you.  Serene Chen

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 8

ely contact the sender if you have received this > message in error. Further, this e-mail may contain viruses and all > reasonable precaution to minimize the risk arising there from is taken by > OnMobile. OnMobile is not liable for any damage sustained by you as a > result

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 10

Thank you for contacting Domio Sports! On Fri, Aug 14 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or, via em

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 6

Thank you for contacting Domio Sports! On Tue, Aug 11 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or, via em

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 7

Thank you for contacting Domio Sports! On Wed, Aug 12 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or, via em

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 9

Thank you for contacting Domio Sports! On Thu, Aug 13 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or, via em

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 5

Thank you for contacting Domio Sports! On Mon, Aug 10 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or, via em

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 4

Thank you for contacting Domio Sports! On Sun, Aug 09 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or, via em

Re: [Sip-implementors] Sip-implementors Digest, Vol 76, Issue 9

Okay, thank you messages czw., 13 sie 2020, 18:00 użytkownik < sip-implementors-requ...@lists.cs.columbia.edu> napisał: > Send Sip-implementors mailing list submissions to > sip-implementors@lists.cs.columbia.edu > > To subscribe or unsubscribe via the World Wide Web, visit > http

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 3

Thanks for contacting Domio Sports! Cheers! On Fri, Aug 07 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or,

[Sip-implementors] Sip-implementors Digest, Vol 76, Issue 3

Thanks for contacting Domio Sports! Cheers! On Sat, Aug 08 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or,

[Sip-implementors] Sip-implementors Digest, Vol 74, Issue 4

Thanks for contacting Domio Sports! Cheers! On Tue, May 12 2020, at 04:00 PM, Send Sip-implementors mailing list submissions to sip-implementors@lists.cs.columbia.edu To subscribe or unsubscribe via the World Wide Web, visit https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors or,

Re: [Sip-implementors] SIP Session Refresh RFC 4028

vity: Internal > > > > -Original Message----- > > From: sip-implementors-boun...@lists.cs.columbia.edu < > sip-implementors-boun...@lists.cs.columbia.edu> On Behalf Of Basu > Chikkalli > > Sent: den 6 augusti 2020 13:22 > > To: sip-implementors >

Re: [Sip-implementors] SIP Session Refresh RFC 4028

Yes, my bad, you are correct ! BR/pj Sensitivity: Internal -Original Message- From: sip-implementors-boun...@lists.cs.columbia.edu On Behalf Of Paul Kyzivat Sent: den 6 augusti 2020 18:05 To: sip-implementors@lists.cs.columbia.edu Subject: Re: [Sip-implementors] SIP Session Refresh

Re: [Sip-implementors] SIP Session Refresh RFC 4028

Chikkalli Sent: den 6 augusti 2020 13:22 To: sip-implementors Subject: [Sip-implementors] SIP Session Refresh RFC 4028 Hi All, A->B2BUA>B A-Party does not support session. no Session_Expires,no Min-SE and no Supported:timer So no

Re: [Sip-implementors] SIP Session Refresh RFC 4028

-implementors-boun...@lists.cs.columbia.edu On Behalf Of Basu Chikkalli Sent: den 6 augusti 2020 13:22 To: sip-implementors Subject: [Sip-implementors] SIP Session Refresh RFC 4028 Hi All, A->B2BUA>B A-Party does not support session. no Session_E

[Sip-implementors] SIP Session Refresh RFC 4028

Hi All, A->B2BUA>B A-Party does not support session. no Session_Expires,no Min-SE and no Supported:timer So no session refresh between A and B2BUA. When B2BUA supports timer. It sends INVITE to B with following details B2BUA---INVITE---

[Sip-implementors] SIP Federation, authentication and interoperability with WebRTC

Hello everyone, I started looking into SIP over the past months because I find it quite interesting protocol, and after I saw a talk from Olle Johansson. I have went through various RFCs and technical documents and I have some questions that might be naive, but due to lack of experience I need som

Re: [Sip-implementors] Sip proxies and REST endpoints

Thanks Get Outlook for Android On Thu, Nov 1, 2018 at 10:58 AM -0700, wrote: Hi I have a question regarding open source SIP proxies, I want to allow data that sits on a REST endpoint to the SIP headers as they flow through the proxy. Currently a Cisco CME is connected to

Re: [Sip-implementors] Sip proxies and REST endpoints

It sounds like Kamailio would do the trick for you, since it has a number of HTTP clients: https://kamailio.org/docs/modules/5.1.x/modules/http_client.html https://kamailio.org/docs/modules/5.1.x/modules/http_async_client.html And even a nice module for parsing JSON: https://kamailio.org/docs/mo

[Sip-implementors] Sip proxies and REST endpoints

Hi I have a question regarding open source SIP proxies, I want to allow data that sits on a REST endpoint to the SIP headers as they flow through the proxy. Currently a Cisco CME is connected to a SIP trunk. Can we have a CME connect to the proxy, and the proxy to the SIP trunk. The SIP proxy

Re: [Sip-implementors] SIP 422 and RFC 4028

out there > >> whose equipment will just send you a reinvite every 15 minutes to check > if > >> you're alive, quite regardless of any SSTs, roles, support for SSTs, > etc. > >> > >> -- Alex > >> > >> -- &

Re: [Sip-implementors] SIP 422 and RFC 4028

e, please forgive typos and brevity. ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors _______ Sip-implementors mailing list Sip-imple

Re: [Sip-implementors] SIP 422 and RFC 4028

Yeah, that's true. It's easily forgot in an applied sense because the mainstream FOSS proxies, e.g. Kamailio, both support dialog state tracking and issuing various kinds of in-dialog DPD requests (e.g. OPTIONS), and even support spoofing BYEs to hang up a dead call if DPD requests go unreplied

Re: [Sip-implementors] SIP 422 and RFC 4028

tc. -- Alex -- Sent via mobile, please forgive typos and brevity. ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors _______ Sip-

Re: [Sip-implementors] SIP 422 and RFC 4028

On 7/3/18 3:53 AM, Alex Balashov wrote: No, it's not illegal to retry a call to the same gateway (in case of 6xx response). Nor is it illegal to reject it. :-) My experience in an applied sense with SSTs (SIP Session Timers) is that they are poorly supported, seemingly due to all the state-ke

Re: [Sip-implementors] SIP 422 and RFC 4028

heck if > you're alive, quite regardless of any SSTs, roles, support for SSTs, etc. > > -- Alex > > -- > Sent via mobile, please forgive typos and brevity. > ___ > Sip-implementors mailing list > Sip-implementors@lis

Re: [Sip-implementors] SIP 422 and RFC 4028

No, it's not illegal to retry a call to the same gateway (in case of 6xx response). Nor is it illegal to reject it. :-) My experience in an applied sense with SSTs (SIP Session Timers) is that they are poorly supported, seemingly due to all the state-keeping involved. Many UAs commit to a refr

[Sip-implementors] SIP 422 and RFC 4028

Hi All, We have noticed that one provider is not reattempting the call with new session-expire value once the call is rejected with 422 Session Interval Too Small. But RFC 4028 doesn't say its mandatory to retry the call by UAC, but isn't a wrong behavior of UAC? I understand that session-expire

[Sip-implementors] SIP ALG detector

Hello, I was trying to use your SIP ALG detector for business purposes at https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors. The link brings up a 404 Error. Do you mind sending me the link to the SIP ALG detector that is not broken? I would very much appreciate it. Thank you! M

Re: [Sip-implementors] SIP Bye before 200 OK for Initial Invite

Hi Dale, Thank you very much. Its clear now. I found one as well under 3261 under Paragraph 15. On Fri, Jun 9, 2017 at 8:04 PM, Dale R. Worley wrote: > NK writes: > > I am facing a strange scenario where my SBC is sending BYE before it > > received 200 OK for initial invite. > > > > A ==

Re: [Sip-implementors] SIP Bye before 200 OK for Initial Invite

NK writes: > I am facing a strange scenario where my SBC is sending BYE before it > received 200 OK for initial invite. > > A = INVITE > B > A <=== 100 Giving a Try B > A <==100 Try B > A <==183 w/SDP==B > A <==183 w/SDP==B > A <=

[Sip-implementors] SIP Bye before 200 OK for Initial Invite

Dear All, I am facing a strange scenario where my SBC is sending BYE before it received 200 OK for initial invite. A = INVITE > B A <=== 100 Giving a Try B A <==100 Try B A <==183 w/SDP==B A <==183 w/SDP==B A <==180 w/SDP

Re: [Sip-implementors] SIP REGISTRTION INFO

her UA in a dialogue. *It SHOULD be the same in each registration from a UA.* *Best Regards* *Rakesh Kumar Mohanty* ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-i

Re: [Sip-implementors] SIP REGISTRTION INFO

Sip-implementors@lists.cs.columbia.edu <mailto:Sip-implementors@lists.cs.columbia.edu> https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors <https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors>

Re: [Sip-implementors] SIP REGISTRTION INFO

) the call id now changed to 12 and as >>> per >>> RFC 3261 it says >>> >>> The first question is, >>> >>> It says 8.1.1.4 Call-ID >>> >>> >>>The Call-ID header field acts as a unique identifier to grou

Re: [Sip-implementors] SIP REGISTRTION INFO

her UA in a dialogue. *It SHOULD be the same in each registration from a UA.* *Best Regards* *Rakesh Kumar Mohanty* ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-i

Re: [Sip-implementors] SIP REGISTRTION INFO

ilto:rak...@gmail.com] > *Sent:* Friday, January 06, 2017 9:59 AM > > *To:* Brett Tate > *Subject:* Re: [Sip-implementors] SIP REGISTRTION INFO > > > > Hi Brett, > > > > Thanks for the response. So here the UAS should handle it instead > rejecting. > > My case

Re: [Sip-implementors] SIP REGISTRTION INFO

> In our case the Call-ID: ee74e9624ebe1844 value us > differ from the previous one so the UAS should update > the binding and accept the REGISTER request instead > rejecting it with 401. 401 is associated with authentication. They have the right to generate 401 if they want. For instance, they

Re: [Sip-implementors] SIP REGISTRTION INFO

field acts as a unique identifier to group together >> a series of messages. It MUST be the same for all requests >>and responses sent by either UA in a dialogue. >> *It SHOULD be the same in each registration from a UA.*

Re: [Sip-implementors] SIP REGISTRTION INFO

ohanty* > ___ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

[Sip-implementors] SIP REGISTRTION INFO

Hi Expert, I need one info for the following below SIP REGISTER scenario. Step 1) UE sent REGISTER with CSeq =1 CallID=11 Step 2) 401 with CSeq =1 CallID=11 Step 3) After challenge UE sent the REGISTER with CSeq =2 CallID=11 Step 4 ) 200OK with same CSeq =2 CallID=11 which is expe

Re: [Sip-implementors] SIP Request URI Domain processing without Port

Varadhan Work writes: > Suppose, > SIP Domain name: *mydomain.org * > SIP Port: *8060* > Question is, > If Req-URI received [by the Registrar] without port number on SIP > server listening port 8060 as follows, > > *REGISTER sip:mydomain.org SIP/2.0* > >

Re: [Sip-implementors] SIP Request URI Domain processing without Port

060 > port routing ? > and > should be manipulated based on received(socket) port number ? > > or > > b) it should be considered as default 5060 and shouldn't be processed for > 8060 port routing ? > > > Thanks & Regards, > Varadhan > _____

[Sip-implementors] SIP Request URI Domain processing without Port

Hello Everyone, I'm trying to figure out proper method for implementing sip-domain routing policy, while using port number different than default 5060. Suppose, SIP Domain name: *mydomain.org * SIP Port: *8060* SIP Client configured with domain name with port number *mydomai

Re: [Sip-implementors] SIP Domain with single part

wor...@ariadne.com (Dale R. Worley) writes: > However, there has > been at least one exception: Fredrik Thulin once managed the nameserver > for .SE and added an MX record for .SE, directed to the host that > handled his e-mail. Thus, was a valid e-mail address for > him. Thinking more about th

Re: [Sip-implementors] SIP Domain with single part

Varadhan Work writes: > Is it legal to have SIP domain name in SIP packet with only one single part > without top level domain name ? In both the general domain name syntax and in RFC 3261, section 25.1, production "hostname" you can see that a domain name with only one label is syntactically *le

Re: [Sip-implementors] SIP Domain with single part

mited by dots, such > as example.com <https://en.wikipedia.org/wiki/Example.com>.* > Which means, can it be specified with one part in SIP ? > > Thanks & Regards, > Varadhan > ___ > Sip-implementors mailing list > Sip

Re: [Sip-implementors] SIP Domain with single part

sts.cs.columbia.edu [mailto:sip- > implementors-boun...@lists.cs.columbia.edu] On Behalf Of Varadhan Work > Sent: Tuesday, January 03, 2017 2:09 AM > To: sip-implementors@lists.cs.columbia.edu > Subject: [Sip-implementors] SIP Domain with single part > > Hello, > > Is it legal

Re: [Sip-implementors] SIP Domain with single part

On 1/3/17 2:08 AM, Varadhan Work wrote: Hello, Is it legal to have SIP domain name in SIP packet with only one single part without top level domain name ? In general domain name syntax *A domain name consists of one or more parts, technically called labels, that are conventionally concatenated

[Sip-implementors] SIP Domain with single part

Hello, Is it legal to have SIP domain name in SIP packet with only one single part without top level domain name ? In general domain name syntax *A domain name consists of one or more parts, technically called labels, that are conventionally concatenated, and delimited by dots, such as example.c

Re: [Sip-implementors] SIP-URI Without Userinfo

On 11/7/16 4:49 AM, Adnan Aziz wrote: Hello Everyone, According to the rfc3261, the userinfo in the sip-uri is optional. If there is no userinfo part then there should not be any "@" sign as well in the sip-uri. I have couple of questions 1) If I am receiving a sip request without userinfo pa

[Sip-implementors] SIP-URI Without Userinfo

Hello Everyone, According to the rfc3261, the userinfo in the sip-uri is optional. If there is no userinfo part then there should not be any "@" sign as well in the sip-uri. I have couple of questions 1) If I am receiving a sip request without userinfo part in the sip-usri, starting with "@" th

Re: [Sip-implementors] SIP session setup

gt; Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

[Sip-implementors] SIP session setup

Hi Everyone, I have couple of basic questions regarding SIP session setup 1) is it compulsory for UAC to REGISTER to SIP server before session setup? 2) does SIP server checks if the UAC is REGISTERED to the SIP server if it has received an INVITE request from a UAC? If yes then how SIP server han

[Sip-implementors] SIP SDP handling with NAT64 setups

Hi,  When a UA which supports IPV6 and IPV4 is connected to an IPV6 network it could be that SIP messages are routed to IPV4 SIP server via NAT64. In which case the SDP will contain IPV6 media line which the IPV4 server should rejectwith 488. In this scenario is it acceptable to make an offer wit

Re: [Sip-implementors] Sip-implementors Digest, Vol 29, Issue 16

Surya writes: > Please let me know what should be the exact call flow when user select > Airplane mode ON when calling to other name. Heh! It's pretty simple: UA | | | | | | | | | | | | I.e., with airplan

Re: [Sip-implementors] Sip-implementors Digest, Vol 29, Issue 16

nd it is *done*.) But it is contrary to 6665, and > (arguably) to 3265. > > Thanks, > Paul > > > Thanks and Regards > > Dheeraj Kumar > > > > Sent from iPhone > > > >> On 26-Jul-2016, at 9:42 AM, pratik patel > wrote: > >

Re: [Sip-implementors] SIP to SS7 Calling Number Information

I would suggest RCF 3398 for SIP to SS7 mappings and other data. Also below for Default SIP-to-SS7 ISUP Cause Codes. Good luck http://www.dialogic.com/webhelp/img1010/10.5.1/webhelp/General_Reference/def_sip-ss7_cc.htm From: discussion-boun...@sipforum.org [mailto:discussion-boun...@sipforu

[Sip-implementors] SIP Register

Hello: I am using android SIP Demo application to connect to my IMS (SBC / P-CSCF). I am able to send initial REGISTER to IMS system but second REGISTER (i.e. for Authorization) contains different Request-URI (i.e. with port 5060). I don’t want to send port 5060 in second REGISTER Request-URI. I

Re: [Sip-implementors] [SIP-IMPLEMENTORS] Query related to Call Transfer

Hi Brett, Thank you for the response, On Mon, Apr 18, 2016 at 2:40 PM, Brett Tate wrote: > > I am reading the RFC5589 related to the Call Transfer. > > > > > In Examples it is mentioned as below > > 6.1. Successful Transfer > > Why in the CSeq Method value is REFER ? > > The RFC contains som

Re: [Sip-implementors] [SIP-IMPLEMENTORS] Query related to Call Transfer

> I am reading the RFC5589 related to the Call Transfer. > In Examples it is mentioned as below > 6.1. Successful Transfer > Why in the CSeq Method value is REFER ? The RFC contains some errors. See errata 1892. https://www.rfc-editor.org/errata_search.php?rfc=5589 ___

[Sip-implementors] [SIP-IMPLEMENTORS] Query related to Call Transfer

Hi All, I am reading the RFC5589 related to the Call Transfer. In Examples it is mentioned as below 6.1 . Successful Transfer Transferor Transferee Transfer ||

Re: [Sip-implementors] SIP INVITE METHOD

    >Some queries regarding, >*) How many contacts can be present in 3XX reponse ..? See SIP RFC 3261 Section 21.3 -> Depends on the response type I guess >*) Other than INVITE method whether in any other method contact header field >is mandatory ? See SIP RFC 3261 table 2 and section 8.1.1

[Sip-implementors] SIP INVITE METHOD

Hi all, Some queries regarding, *) How many contacts can be present in 3XX reponse ..? *) Other than INVITE method whether in any other method contact header field is mandatory ? *) If the first element in the route set is a loose router then client should take the first element as a

Re: [Sip-implementors] Sip message method has a dialog or not

plementors@lists.cs.columbia.edu Subject: [Sip-implementors] Sip message method has a dialog or not Hi all, I have one doubt whether sip message method has a dialog or not ? If it is so what is the purpose of creating a dialog in MESSAGE Method _

Re: [Sip-implementors] Sip message method has a dialog or not

; implementors-boun...@lists.cs.columbia.edu] On Behalf Of Karthik.v > Sent: Tuesday, January 05, 2016 10:40 AM > To: sip-implementors@lists.cs.columbia.edu > Subject: [Sip-implementors] Sip message method has a dialog or not > > Hi all, > >I have one doubt whether sip message me

[Sip-implementors] Sip message method has a dialog or not

Hi all, I have one doubt whether sip message method has a dialog or not ? If it is so what is the purpose of creating a dialog in MESSAGE Method Thanks, Karthik V ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu

Re: [Sip-implementors] SIP

AS. The route set is used by that UA to construct requests within the dialog. The details are given in RFC 3261.) Dale ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columb

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