Re: [Sip-implementors] RFC 5407 error?

2011-04-12 Thread Peter Krebs
First of all, many thanks for both answers. As for your modified example, I think this situation (UAS receiving an ACK in mortal state, state change caused by BYE received from UAC) is discussed in sec. 3.1.6 of RFC 5407 (with the sole difference that the first ACK is lost, not reordered as in

Re: [Sip-implementors] RTP/AVP with crypto attribute

2011-04-12 Thread Saúl Ibarra Corretgé
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

Re: [Sip-implementors] Phone-context header

2011-04-12 Thread Sumit Jindal
RFC 3261. spot on. Regards, Sumit Jindal On Tue, Apr 12, 2011 at 3:04 PM, Manoj Priyankara [TG] mano...@suntel.lkwrote: Dear All, Please help me understand the use of phone-context field in the From header of the INVITE. Is it globally significant? How to process the INVITE coming to a

Re: [Sip-implementors] Phone-context header

2011-04-12 Thread Attila Sipos
Hi Manoj, I did ask about phone-context a while ago: https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-March/0188 35.html RFC 3966 might help too: http://www.rfc-editor.org/rfc/rfc3966.txt Regards Attila -Original Message- From:

Re: [Sip-implementors] Phone-context header

2011-04-12 Thread Manoj Priyankara [TG]
Thanks all. -Original Message- From: Attila Sipos [mailto:attila.si...@vegastream.com] Sent: Tuesday, April 12, 2011 3:51 PM To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Phone-context header Hi Manoj, I did ask about phone-context

Re: [Sip-implementors] RFC 5407 error?

2011-04-12 Thread Pekka Pessi
Hello, On Mon, 2011-04-11 at 12:50 +0200, ext Peter Krebs wrote: While perusing the race condition examples in RFC 5407 I noticed what seems to be an inconsistency in regard to RFC 3261. The example in sec. 3.2.4 depicts the callee (Bob) sending a BYE request immediately after the 200,

[Sip-implementors] rport is useless in TCP/SCTP

2011-04-12 Thread Iñaki Baz Castillo
Hi, RFC 3581 defines the usage of ;rport Via param in order for a server to send UDP responses to the request source port (rather than the port indicated in Via sent-by). But this is completely useless in SIP over TCP as, per RFC 3261, responses are primary sent over the same connection in which

Re: [Sip-implementors] RFC 5407 error?

2011-04-12 Thread Brett Tate
While sending BYE for the initial INVITE is prohibited by RFC 3261 before an ACK is received, I guess this does not hold for RE-INVITEs since the dialog is already established. Is this correct, as re-invites are not mentioned at all in sec. 15? RFC 3261 does not generically require the

[Sip-implementors] SIP TCP: Retransmission over a different TCP connection, where to reply?

2011-04-12 Thread Iñaki Baz Castillo
Hi, a server receives a SIP request over TCP and creates a server transaction. While processing it, the client sends a request retransmission but using a different TCP connection. Note that request matching rules on RFC 3261 - 17.2.3 are passed (same branch, method and Via sent-by), even if the

Re: [Sip-implementors] SIP TCP: Retransmission over a different TCP connection, where to reply?

2011-04-12 Thread Worley, Dale R (Dale)
From: sip-implementors-boun...@lists.cs.columbia.edu [sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Iñaki Baz Castillo [i...@aliax.net] Now the question is: where should the server re-send the last replied response? should it send it over

Re: [Sip-implementors] No Ringback in 183 SDP

2011-04-12 Thread Randell Jesup
3 apr 2011 kl. 13.23 skrev Iñaki Baz Castillo: 2011/3/31 Olle E. Johansson o...@edvina.net: If you are sending only ringback, I would recommend sending 180 with SDP instead of 183. If you're sending 183, I can't move my state machine to ringing state, which would help a lot of 3rd party

Re: [Sip-implementors] Modfying ports and IP address in SDP.

2011-04-12 Thread Randell Jesup
From: Jaiswal, Sanjiv 2)The is onging audio session between the agents . Now the terminating end wants to change from audio to video. Termination generates the reinvite with addition m lines for new audio and video and previous mline port set to zero (deletion of media). Peer on receiving the

Re: [Sip-implementors] SIP TCP: Retransmission over a different TCP connection, where to reply?

2011-04-12 Thread Iñaki Baz Castillo
2011/4/12 Worley, Dale R (Dale) dwor...@avaya.com: Clearly it can use either connection, as it can pretend it did not receive the message over the other connection! Humm, I dont' understand, obviously the server did receive the request from the first connection (if not it wouldn't match the

Re: [Sip-implementors] SIP TCP: Retransmission over a different TCP connection, where to reply?

2011-04-12 Thread Worley, Dale R (Dale)
From: Iñaki Baz Castillo [i...@aliax.net] 2011/4/12 Worley, Dale R (Dale) dwor...@avaya.com: Clearly it can use either connection, as it can pretend it did not receive the message over the other connection! Humm, I dont' understand, obviously the

Re: [Sip-implementors] No Ringback in 183 SDP

2011-04-12 Thread Paul Kyzivat
On 4/12/2011 12:21 PM, Randell Jesup wrote: 3 apr 2011 kl. 13.23 skrev Iñaki Baz Castillo: 2011/3/31 Olle E. Johanssono...@edvina.net: If you are sending only ringback, I would recommend sending 180 with SDP instead of 183. If you're sending 183, I can't move my state machine to ringing