[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] 18x response after OA complete?

9:36 > yes there can be more 18X and PRACK > based OA sequence, in case of call forking scenario. > > If there is no call forking it should be UPDATE that need > to be used for modification of offer before 2XX is > receieved. > > Regards > Sunil Verma > > &

Re: [Sip-implementors] 18x response after OA complete?

-Original Message- > From: sip-implementors-boun...@lists.cs.columbia.edu > [mailto:sip-implementors-boun...@lists.cs.columbia.edu] > On Behalf Of > Nauman Sulaiman > Sent: Wednesday, June 29, 2011 4:16 AM > To: sip-implementors@lists.cs.columbia.edu > Subject: [Sip-implementors]

[Sip-implementors] 18x response after OA complete?

Hi, After OA is complete is it possible for UAS to send more 18x responses (with no offer or answer of course). UAC UAS |INVITE-->| | | |<1xx (o)-| - provisional responses with REQUIRE 100rel | |

[Sip-implementors] PRACK not received, should callee end the call?

Hi, In this scenario UA                Server <-     INVITE 100rel required ->     180 (o) <-     PRACK (a) delayed User answers phone >   4XX ??     In the above scenario, if UA generates ringing on receipt of the INVITE

[Sip-implementors] Answer in 200OK following answer in 18X rel

Hi, In the flow below UAC UAS > INVITE(0) require 100rel < 183 rel (a) -> PRACK <- 200OK <- 200K(a) ?? --> ACK after sending a

[Sip-implementors] Reinvite Offer Answer - can previous negotiated codec change?

Hi, Just wondering what is preferred here? UA1 offers PCMU, PCMA, G729 (PCMU most preferred) to UA2 UA2 answers with PCMU This completes OA and PCMU is negotiated codec Then UA2 is put on hold and then unheld UA2 sends in REINVITE (Unhold) offer G729,PCMA,PCMU (G729 most preferred) As G

[Sip-implementors] X-nat:0 attribute in SDP

Hi, I have seen some UAs use this session attribute a= X-nat:0 in SDP. Can someone please explain what this is? Searched the web but could not find anything. Thanks ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists

[Sip-implementors] Is PRACK and UPDATE mandatory in some use cases

Hi Some UAs do not support PRACK or UPDATE and still work fine with various Proxies, PBX, Softswitches etc. Are there any scenarios or well known Proxies,Softswitches that require mandatory PRACK and UPDATE support for a UA to be interoperable? Or maybe some features will not be available if su

[Sip-implementors] Codec renegotiation failure on REFER

Hi, what is the correct response for the follows If codec negotiation fails on initial offer/answer exchange the usual response is 488 Not Acceptable. If it fails during the REINVITE stage I have seen UAs send 500 Internal error. Is this correct or should it still be 488? Thanks ___

Re: [Sip-implementors] Broadworks Reinvite issue

AC is already in hold. And > also, earlier for a=sendonly, UAC had got a=inactive. > > -Somesh > > -Original Message- > From: sip-implementors-boun...@lists.cs.columbia.edu > [mailto:sip-implementors-boun...@lists.cs.columbia.edu] > On Behalf Of ext Nauman S

[Sip-implementors] Broadworks Reinvite issue

Hi, Scenario is as follows UAC makes call to UAS over Broadworks B2BUA and then UAC goes on hold. Broadworks periodically issues Session Audit with SDP version number unchanged, this is for UAC and UAS which do not support session timers or UPDATE method. UAC ---> Broadworks

Re: [Sip-implementors] SIP flow for BYE sent for early dialog

gt; indeed a dialog > formed by the INVITE transaction. > > On Tuesday, 22 February, 2011 06:47 AM, Nauman Sulaiman > wrote: > > Hi, If the caller sends a BYE for an early dialog what > should the callee send as response to INVITE. I understand a > 200K would be sent

[Sip-implementors] SIP flow for BYE sent for early dialog

Hi, If the caller sends a BYE for an early dialog what should the callee send as response to INVITE. I understand a 200K would be sent for the BYE. caller ---> Callee INVITE caller <--- Callee 100 Trying caller -> Callee

[Sip-implementors] Is Replaces header allowed in ReInvite?

Hi, I read RFC 3891 but am unsure whether replaces header is allowed in ReInvite or must it always be in an initial Invite. If it is allowed does this actually happen in practice. Thanks ___ Sip-implementors mailing list Sip-implementors@list

Re: [Sip-implementors] REFER RURI destination

n > To: nauman762-h...@yahoo.co.uk > Cc: sip-implementors@lists.cs.columbia.edu > Date: Monday, 7 February, 2011, 9:23 > 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 co

[Sip-implementors] REFER RURI destination

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 B contains a Refer-To header with the public address of A. If B s

[Sip-implementors] Do all URI parameters in contact need to be copied to RURI?

Hi, If the contact header in 200OK to initial INVITE contains URI parameters (say several) do all these need to be copied into 1) the ACK to the 200 OK 2) any subsequent RE-INVITEs or just the transport parameter if present needs to be copied. I have seen several UA's leave them off. Is it opt

[Sip-implementors] Routing incoming INVITE on proxy

Hi, How to solve the following problem on proxy with SIP UA > Proxy --> Voxalot, AntiSIP others UA ,registers with public IP. Proxy inserts path header. Hi, Some VoiP providers change the contact header sent by UA in REGISTER request to the ip address that the request wa

[Sip-implementors] Format of Accept header in 415 response

Hi, if i send 415 response and wish to inform that only GSM codec is acceptable how do i format the Accept header? Accept : audio GSM ? How to do thanks? ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.c

[Sip-implementors] Error response for OA failure

Hi, For a Re-invite scenario If the OA process fails due to example a UAS sending a 200OK without a body to an INVITE with a body from UAC. What is the correct response of the UAC? Thanks ___ Sip-implementors mailing list Sip-implementors@l

[Sip-implementors] Abort OA after sending 491?

Hi, in a typical case where re-invites cross and UA1 sends a 491 but does not receive one. Should UA1 still abort its local OA processing or continue to wait for a final response? UA1 - Reinvite -->UA2 UA1 <-Reinvite - UA2 UA1 - 491 -> UA2 This can aris

[Sip-implementors] Registrar/Proxy server and contact header in request

Hi, We have a case where we send a REGISTER request from a UA to a REGISTRAR via a proxy. The Registrar in the 200OK response populates the contact header with the address from which it received the request ie that of the proxy. It ignores the contact header( which had private ip) in the Reque

Re: [Sip-implementors] Is Replaces header in Invite still used much?

edu] > On Behalf Of ext > M. Ranganathan > Sent: Monday, October 25, 2010 8:11 AM > To: nauman762-h...@yahoo.co.uk > Cc: sip-implementors@lists.cs.columbia.edu > Subject: Re: [Sip-implementors] Is Replaces header in > Invite still used > much? > > On Mon, Oct 25, 2010 at

[Sip-implementors] Is Replaces header in Invite still used much?

Hi, i have examined some SIP traces of attended transfers using some PBX's such as Asterisk and 3CX using X-Lite softphones and am finding that there are no Replaces headers in the Invites to the transfer target. So where is the Replaces header in Invites used these days and how could i set up so

[Sip-implementors] Proxy 302 redirect violating spec??

Hi , we have a set up where we are using a SIP proxy( OpenSIPS) to handle a 302 redirect locally but it doesn't seem to be working with voip providers here is the call flow. Is it legal, if so we'll bother you with a SIP trace INVITE INVITE VoipFone Server ->

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

Hi , we have a set up where we are using a SIP proxy( OpenSIPS) to handle a 302 redirect locally but it doesn't seem to be working with voip providers here is the call flow. Is it legal, if so we'll bother you with a SIP trace                  INVITE             INVITE VoipFone Server ->

[Sip-implementors] Design Problem with mobile SIP endpoint

Hi, we have the following design requirement.  We need to have a 'home network' which acts as end user agent temporarily for some mobile user agents. The home network will register for these user agents with a voip service provider. Incoming calls will be routed to the 'home network' . The mobil