Re: [Sip-implementors] multiple 100 prov responses in the same invitetransaction

2009-09-15 Thread Tarun2 Gupta
Hi Amy The UAS can send any number of 100 Trying responses. See the following excerpt from RFC 3261. 17.2.1 INVITE Server Transaction When a server transaction is constructed for a request, it enters the "Proceeding" state. The server transaction MUST generate a 100 (Trying) response

Re: [Sip-implementors] overloading static RTP payload types with dynamic payload types

2009-09-15 Thread Dale Worley
On Mon, 2009-08-24 at 08:26 -0700, Amy Hwang wrote: > Let's say SDP is offered with the following media line: > > m=audio 12345 RTP/AVP 97 > a=rtpmap:97 PCMU/8000 > > That is, a dynamic payload type of 97 is being used to represent > PCMU/8000, which is also mapped to the static payload type of 0

Re: [Sip-implementors] Registrar: Contact matching decisions if NAT fails

2009-09-15 Thread Dale Worley
On Tue, 2009-09-08 at 23:24 +0200, Thomas Gelf wrote: > The problem we were talking about are clients filling the registrars > location database with multiple (usually in the range of 3-30) different > records. Same UAC, same Call-ID, same username, same (official) IP but > changing port in their C

Re: [Sip-implementors] Contact mismatch in 200 resp for Register req

2009-09-15 Thread Dale Worley
On Sun, 2009-08-30 at 12:20 +0530, vijay wrote: > Hi, > If we get different contact in 200 response for Register request, Is it > successful registration? > If yes, Can we consider the expires(for reg refresh) value present > in the Contact? > If no, Please suggest, what should we d

Re: [Sip-implementors] multiple 100 prov responses in the same invite transaction

2009-09-15 Thread IƱaki Baz Castillo
2009/9/15 Amy Hwang : > > Hello, > > In RFC 3261 (or 3262 if applicable), is it legal for a UAS to send multiple > 100 provisional responses within the same INVITE transaction? It seems to be > the common practice to send at most one, but I can't find anything explicitly > disallowing sending mo

[Sip-implementors] multiple 100 prov responses in the same invite transaction

2009-09-15 Thread Amy Hwang
Hello, In RFC 3261 (or 3262 if applicable), is it legal for a UAS to send multiple 100 provisional responses within the same INVITE transaction? It seems to be the common practice to send at most one, but I can't find anything explicitly disallowing sending more than one 100 response. Thanks,

Re: [Sip-implementors] Session Expires with less than 90 seconds

2009-09-15 Thread Thomas Gelf
Bhanu K S (bhks) wrote: > I believe 422 can be send only by UAS not UAC. > If UAC receives 200 OK with Session-Expires as 10 seconds(which is less > than 90 default), it is against RFC. > So it is UAC implementation either it can continue to be in the call or > disconnect. That's correct, I didn

Re: [Sip-implementors] Session Expires with less than 90 seconds

2009-09-15 Thread Dushyant Dhalia
The UAC may terminate the call with BYE and include Reason header in it. Dushyant P S Dhalia vijay wrote: Hi, The caller sends INVITE with Session-Expires as 1800. Min-SE expires is not present. The callee sends 200 Ok with Session-Expires as 10 seconds. As per RFC, if Mi

Re: [Sip-implementors] Session Expires with less than 90 seconds

2009-09-15 Thread Tarun2 Gupta
Hi Vijay IMO, the caller should reject a 200 OK response with such a small session refresh interval. Excerpts from RFC 4028 Session Timer in SIP 4. Session-Expires Header Field Definition Small session intervals can be destructive to the network. They cause excessive messaging traffic

Re: [Sip-implementors] Session Expires with less than 90 seconds

2009-09-15 Thread Bhanu K S (bhks)
I believe 422 can be send only by UAS not UAC. If UAC receives 200 OK with Session-Expires as 10 seconds(which is less than 90 default), it is against RFC. So it is UAC implementation either it can continue to be in the call or disconnect. -Bhanu -Original Message- From: sip-implementor

Re: [Sip-implementors] [Sip-Implementors] DTMF using SIP INFO

2009-09-15 Thread Avasarala Ranjit-A20990
Hi INFO method is defined in RFC 2976. But yes "dtmf" and "dtmf-relay" are not standard defined mime types. No there is no registered MIME type for DTMF. Regards Ranjit -Original Message- From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip-implementors-boun...@lists.cs.c

Re: [Sip-implementors] Session Expires with less than 90 seconds

2009-09-15 Thread Thomas Gelf
vijay wrote: > The caller sends INVITE with Session-Expires as 1800. Min-SE expires is > not present. > The callee sends 200 Ok with Session-Expires as 10 seconds. > > As per RFC, if Min-SE is not present, the default is 90 seconds. > In this error case, can the user stop the call? Pl clarify.

Re: [Sip-implementors] [Sip-Implementors] DTMF using SIP INFO

2009-09-15 Thread Shefali Dutta
Hi, Thanks for your reply. The link below gives two MIME types application/dtmf and application/dtmf-relay but additionally gives the Note -> "Note: dtmf-relay or dtmf are not yet IANA registered application mime types" Is there some other registered MIME type for DTMF? Rgds, Shefali -O

Re: [Sip-implementors] [Sip-Implementors] DTMF using SIP INFO

2009-09-15 Thread Shefali Dutta
Hi, Thanks for your reply. RFC 2833 does not talk of SIP INFO method. It only talks of carrying the DTMF in RTP payload? Is there any other reference or I have not understood 2833 correctly? Rgds, Shefali -Original Message- From: Avasarala Ranjit-A20990 [mailto:ran...@motorola.com] S

[Sip-implementors] [Sip-Implementors] DTMF using SIP INFO

2009-09-15 Thread Shefali Dutta
Hi All, How DTMF is carried using SIP INFO method? I found some examples using Content type application/dtmf and application/dtmf-relay. Which is the reference RFC for these content types? Rgds, Shefali "DISCLAIMER: This message is proprietary to Aricent and

Re: [Sip-implementors] RFC 2833 Telephone-Event RTP Payload Type

2009-09-15 Thread Attila Sipos
If A says 97 and B says 101, B must send using payload 97 A must send using payload 101 They work like ports: If A says port 1 and B says port 2 B must send to port 1 A must send to port 2 In the case of RTP, if a particular codec was referenced with a specific payload

[Sip-implementors] Session Expires with less than 90 seconds

2009-09-15 Thread vijay
Hi, The caller sends INVITE with Session-Expires as 1800. Min-SE expires is not present. The callee sends 200 Ok with Session-Expires as 10 seconds. As per RFC, if Min-SE is not present, the default is 90 seconds. In this error case, can the user stop the call? Pl