Hi, Thanks for the reply. I have tried with various Servers that support Conferencing like Communigate Pro, Quorum etc. All these have one thing in common. My UA sends Invite to the conference server and the server has to send back isfocus parameter in the contact header field of 200 Ok response which none of the servers are found supporting. So I am still looking for a conferencing server that supports it and the RFC 4579.
Any ideas? Thanks again, Padmaja On Mon, Sep 22, 2008 at 6:33 PM, < [EMAIL PROTECTED]> wrote: > Send Sip-implementors mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Sip-implementors digest..." > > > Today's Topics: > > 1. Re: Should 'ptime' value be multiple of 5 or 10 (Sachin Rastogi) > 2. Re: SipConnect Registration/Authentication > (Rastogi, Vipul (Vipul)) > 3. Re: Should 'ptime' value be multiple of 5 or 10 (Schwarz Albrecht) > 4. Re: Should 'ptime' value be multiple of 5 or 10 (Aneesh Naik) > 5. Re: SIP conference server for testing (Padmaja) > 6. Re: SIP conference server for testing (Minakshi Anand) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 22 Sep 2008 17:14:44 +0530 > From: "Sachin Rastogi" <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] Should 'ptime' value be multiple of 5 > or 10 > To: "Aneesh Naik" <[EMAIL PROTECTED]> > Cc: [email protected] > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Thanks Aneesh,Alex for quick reply. > > > Sachin > > On Mon, Sep 22, 2008 at 3:03 PM, Aneesh Naik <[EMAIL PROTECTED]> > wrote: > > > Yes, as explained by alex the ptime depend on the frame size... > > The frame size for g.711 is 5ms, for g.729 is 10ms and for g.723 it is 30 > > ms. > > So the ptime for 711 and 729 can be 10, 20, 30... and for 723 it can be > > 30,60.... > > > > -Aneesh > > > > On Mon, Sep 22, 2008 at 1:50 PM, Alex Balashov < > [EMAIL PROTECTED] > > >wrote: > > > > > There are no drafts or recommendations mandating that ptime be a > > > particular value because the desired packetisation duration is closely > > > related to the frame size and other characteristics of particular > codecs. > > > > > > However, almost no media gateway equipment out there will support > > > anything other than the usual 10-30 ms for the usual suspects - > > > G.711u/A, G.729A, etc. > > > > > > There are certain recommendations or mandates for ptime values based on > > > particular codecs. For instance, 20 ms is recommended when using Speex > > > (http://www.speex.org/docs/manual/speex-manual/node14.html): > > > > > > --- > > > > > > 4.1.1. Registration of media type audio/speex > > > > > > Media type name: audio > > > > > > Media subtype name: speex > > > > > > Required parameters: > > > > > > None > > > > > > Optional parameters: > > > > > > ptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > > > > > maxptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > > > > > > > > > > > Sachin Rastogi wrote: > > > > > > > Hi All, > > > > Is there any rfc/draft or recommendation which suggests that 'ptime' > > > > value should multiple of 5 or 10? Can I use 'ptime' value as 11 or 12 > > or > > > 18 > > > > or 19? > > > > > > > > > > > > > > > > Thanks > > > > Sachin Rastogi > > > > _______________________________________________ > > > > Sip-implementors mailing list > > > > [email protected] > > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > > > -- > > > Alex Balashov > > > Evariste Systems > > > Web : http://www.evaristesys.com/ > > > Tel : (+1) (678) 954-0670 > > > Direct : (+1) (678) 954-0671 > > > Mobile : (+1) (706) 338-8599 > > > _______________________________________________ > > > Sip-implementors mailing list > > > [email protected] > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > > > > > -- > > Aneesh > > _______________________________________________ > > Sip-implementors mailing list > > [email protected] > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > ------------------------------ > > Message: 2 > Date: Mon, 22 Sep 2008 19:53:41 +0800 > From: "Rastogi, Vipul (Vipul)" <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] SipConnect Registration/Authentication > To: "Vivek Batra" <[EMAIL PROTECTED]>, > <[email protected]> > Message-ID: > < > [EMAIL PROTECTED]> > Content-Type: text/plain; charset="US-ASCII" > > >>>>My Query --> Regarding point 3, should we not send the DID number in > the FROM field so that the called party can *identify* the actual > caller. > [VR]This will be done by P-Preferred-Identity (child id), in case of > anonymous, P-Preferred-Identity has child id, P-Asserted-Identity has > parent id. > > >>>>What will be the Request-URI and TO field when IPPBX receives the > incoming INVITE? Should we expect the parent number viz 12304 in > Request-URI and DDI number in TO field viz (12302 for example) so that > PBX can identify the called party on the basis of TO field. > [VR] Right but I have also seen reverse. So this is inter-op issue. > > Vipul Rastogi > > ________________________________ > > From: Vivek Batra [mailto:[EMAIL PROTECTED] > Sent: Friday, September 19, 2008 4:39 PM > To: Rastogi, Vipul (Vipul) > Subject: Re: [Sip-implementors] SipConnect Registration/Authentication > > > > Hi Vipul, > > This mail is in concern with your posting on SIP Implementors about DID. > I just wanted to confirm what you said: > > > > 1. Suppose I have purchased 10 DDI numbers from BroadWorks. Parent > number is 12301 and DDI numbers are 12301 to 12310. > 2. Now I need to send the REGISTER request only for parent id viz > 12301 (shall be present in FROM field). > 3. When any user of IPPBX makes an OG call to BroadWorks server, > PBX shall send parent id in the FROM field viz 12301. > > > > My Query --> Regarding point 3, should we not send the DID number in the > FROM field so that the called party can *identify* the actual caller. > > > > 4. What will be the Request-URI and TO field when IPPBX receives > the incoming INVITE? Should we expect the parent number viz 12304 in > Request-URI and DDI number in TO field viz (12302 for example) so that > PBX can identify the called party on the basis of TO field. > > > > I would be much pleased if you comment on point 3 and point 4. > > > > Best Regards, > > Vivek Batra > > > > > > > > Date: Thu, 18 Sep 2008 17:38:41 +0800 > > From: "Rastogi, Vipul (Vipul)" <[EMAIL PROTECTED]> > > Subject: Re: [Sip-implementors] SipConnect Registration/Authentication > > Questions > > To: <[EMAIL PROTECTED]>, <[email protected]> > > Message-ID: > > > <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset="US-ASCII" > > > > Most of Core Service Provider VOIP Platforms supports SIPConnect option > > 1 and option 2 but they prefer option 2. BroadWorks is one of them. In > > option 2, we just register with main Id (called Parent ID) and every out > > going INVITE has Parent Id in 'FROM' header. Option 2 behaves like > > single endpoint to ISP S/W. > > > > - > > Vipul Rastogi > > > > > > ------------------------------ > > Message: 3 > Date: Mon, 22 Sep 2008 14:13:18 +0200 > From: "Schwarz Albrecht" <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] Should 'ptime' value be multiple of 5 > or 10 > To: "Sachin Rastogi" <[EMAIL PROTECTED]>, "Aneesh Naik" > <[EMAIL PROTECTED]> > Cc: [email protected] > Message-ID: > < > [EMAIL PROTECTED]> > > Content-Type: text/plain; charset="iso-8859-1" > > > > The frame size for g.711 is 5ms > > G.711 has a frame size of 125 ?s in my understanding. > See ITU-T G.711. > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On > > Behalf Of Sachin Rastogi > > Sent: Montag, 22. September 2008 13:45 > > To: Aneesh Naik > > Cc: [email protected] > > Subject: Re: [Sip-implementors] Should 'ptime' value be > > multiple of 5 or 10 > > > > Thanks Aneesh,Alex for quick reply. > > > > > > Sachin > > > > On Mon, Sep 22, 2008 at 3:03 PM, Aneesh Naik > > <[EMAIL PROTECTED]> wrote: > > > > > Yes, as explained by alex the ptime depend on the frame size... > > > The frame size for g.711 is 5ms, for g.729 is 10ms and for > > g.723 it is > > > 30 ms. > > > So the ptime for 711 and 729 can be 10, 20, 30... and for > > 723 it can > > > be 30,60.... > > > > > > -Aneesh > > > > > > On Mon, Sep 22, 2008 at 1:50 PM, Alex Balashov > > > <[EMAIL PROTECTED] > > > >wrote: > > > > > > > There are no drafts or recommendations mandating that ptime be a > > > > particular value because the desired packetisation duration is > > > > closely related to the frame size and other > > characteristics of particular codecs. > > > > > > > > However, almost no media gateway equipment out there will support > > > > anything other than the usual 10-30 ms for the usual suspects - > > > > G.711u/A, G.729A, etc. > > > > > > > > There are certain recommendations or mandates for ptime > > values based > > > > on particular codecs. For instance, 20 ms is recommended > > when using > > > > Speex > > > > (http://www.speex.org/docs/manual/speex-manual/node14.html): > > > > > > > > --- > > > > > > > > 4.1.1. Registration of media type audio/speex > > > > > > > > Media type name: audio > > > > > > > > Media subtype name: speex > > > > > > > > Required parameters: > > > > > > > > None > > > > > > > > Optional parameters: > > > > > > > > ptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > > > > > > > maxptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > > > > > > > > > > > > > > > Sachin Rastogi wrote: > > > > > > > > > Hi All, > > > > > Is there any rfc/draft or recommendation which suggests > > that 'ptime' > > > > > value should multiple of 5 or 10? Can I use 'ptime' > > value as 11 or > > > > > 12 > > > or > > > > 18 > > > > > or 19? > > > > > > > > > > > > > > > > > > > > Thanks > > > > > Sachin Rastogi > > > > > _______________________________________________ > > > > > Sip-implementors mailing list > > > > > [email protected] > > > > > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > > > > > > -- > > > > Alex Balashov > > > > Evariste Systems > > > > Web : http://www.evaristesys.com/ > > > > Tel : (+1) (678) 954-0670 > > > > Direct : (+1) (678) 954-0671 > > > > Mobile : (+1) (706) 338-8599 > > > > _______________________________________________ > > > > Sip-implementors mailing list > > > > [email protected] > > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > > > > > > > > > > -- > > > Aneesh > > > _______________________________________________ > > > Sip-implementors mailing list > > > [email protected] > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > _______________________________________________ > > Sip-implementors mailing list > > [email protected] > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > ------------------------------ > > Message: 4 > Date: Mon, 22 Sep 2008 18:14:39 +0530 > From: "Aneesh Naik" <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] Should 'ptime' value be multiple of 5 > or 10 > To: "Schwarz Albrecht" <[EMAIL PROTECTED]> > Cc: [email protected] > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > 125 ?s is not the frame size of G.711. It is the time sampling rate. > > 8000 samples in 1 sec. > 1 sample in 125 ?s. > > Number of such samples are combined to form a frame of data. > This frame size for G.711 is 5ms. > > Hope this clears your understanding. > > Thanks, > Aneesh > > On Mon, Sep 22, 2008 at 5:43 PM, Schwarz Albrecht < > [EMAIL PROTECTED]> wrote: > > > > > The frame size for g.711 is 5ms > > > > G.711 has a frame size of 125 ?s in my understanding. > > See ITU-T G.711. > > > > > > > -----Original Message----- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On > > > Behalf Of Sachin Rastogi > > > Sent: Montag, 22. September 2008 13:45 > > > To: Aneesh Naik > > > Cc: [email protected] > > > Subject: Re: [Sip-implementors] Should 'ptime' value be > > > multiple of 5 or 10 > > > > > > Thanks Aneesh,Alex for quick reply. > > > > > > > > > Sachin > > > > > > On Mon, Sep 22, 2008 at 3:03 PM, Aneesh Naik > > > <[EMAIL PROTECTED]> wrote: > > > > > > > Yes, as explained by alex the ptime depend on the frame size... > > > > The frame size for g.711 is 5ms, for g.729 is 10ms and for > > > g.723 it is > > > > 30 ms. > > > > So the ptime for 711 and 729 can be 10, 20, 30... and for > > > 723 it can > > > > be 30,60.... > > > > > > > > -Aneesh > > > > > > > > On Mon, Sep 22, 2008 at 1:50 PM, Alex Balashov > > > > <[EMAIL PROTECTED] > > > > >wrote: > > > > > > > > > There are no drafts or recommendations mandating that ptime be a > > > > > particular value because the desired packetisation duration is > > > > > closely related to the frame size and other > > > characteristics of particular codecs. > > > > > > > > > > However, almost no media gateway equipment out there will support > > > > > anything other than the usual 10-30 ms for the usual suspects - > > > > > G.711u/A, G.729A, etc. > > > > > > > > > > There are certain recommendations or mandates for ptime > > > values based > > > > > on particular codecs. For instance, 20 ms is recommended > > > when using > > > > > Speex > > > > > (http://www.speex.org/docs/manual/speex-manual/node14.html): > > > > > > > > > > --- > > > > > > > > > > 4.1.1. Registration of media type audio/speex > > > > > > > > > > Media type name: audio > > > > > > > > > > Media subtype name: speex > > > > > > > > > > Required parameters: > > > > > > > > > > None > > > > > > > > > > Optional parameters: > > > > > > > > > > ptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > > > > > > > > > maxptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > > > > > > > > > > > > > > > > > > > Sachin Rastogi wrote: > > > > > > > > > > > Hi All, > > > > > > Is there any rfc/draft or recommendation which suggests > > > that 'ptime' > > > > > > value should multiple of 5 or 10? Can I use 'ptime' > > > value as 11 or > > > > > > 12 > > > > or > > > > > 18 > > > > > > or 19? > > > > > > > > > > > > > > > > > > > > > > > > Thanks > > > > > > Sachin Rastogi > > > > > > _______________________________________________ > > > > > > Sip-implementors mailing list > > > > > > [email protected] > > > > > > > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > > > > > > > > > -- > > > > > Alex Balashov > > > > > Evariste Systems > > > > > Web : http://www.evaristesys.com/ > > > > > Tel : (+1) (678) 954-0670 > > > > > Direct : (+1) (678) 954-0671 > > > > > Mobile : (+1) (706) 338-8599 > > > > > _______________________________________________ > > > > > Sip-implementors mailing list > > > > > [email protected] > > > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > > > > > > > > > > > > > > > -- > > > > Aneesh > > > > _______________________________________________ > > > > Sip-implementors mailing list > > > > [email protected] > > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > _______________________________________________ > > > Sip-implementors mailing list > > > [email protected] > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > > > > > -- > Aneesh > > > ------------------------------ > > Message: 5 > Date: Mon, 22 Sep 2008 18:15:00 +0530 > From: Padmaja <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] SIP conference server for testing > To: [email protected], [EMAIL PROTECTED] > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > Thanks a lot for clearing that doubt. Do you know of any free conference > server supporting the RFC 4579 "Call Control - Conferencing for User > Agents"? I have been searching for a long time but no luck so far. Please > let me know if you know of any. > > Thanks, > Padmaja > > > > Message: 2 > Date: Mon, 15 Sep 2008 15:48:31 +0200 > From: " I?aki Baz Castillo " <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] SIP conference server for testing > Cc: [email protected] > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=UTF-8 > > 2008/9/15, [EMAIL PROTECTED] <[EMAIL PROTECTED]>: > > Asterisk is one open source SIP PBX, which supports conferencing. You > > can try that... > > All SIP Asterisk imlpementation is 100% server side without client > knowledge (in fact, client calls to an Asterisk extension and Asterisk > inserts it into a media conference). The client knows *nothing* (via > SIP I mean) about other participants, and of course it doesn't use SIP > RFC's related to SIP native conference . > > Note that the original question asked for a server supporting > "isfocus" feature (present in RFC 4579 "Call Control - Conferencing > for User Agents"). > > > ------------------------------ > > Message: 6 > Date: Mon, 22 Sep 2008 18:32:56 +0530 > From: Minakshi Anand <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] SIP conference server for testing > To: Padmaja <[EMAIL PROTECTED]>, > "[email protected]" > <[email protected]>, "[EMAIL PROTECTED]" > <[EMAIL PROTECTED]> > Message-ID: > < > [EMAIL PROTECTED]> > > Content-Type: text/plain; charset="us-ascii" > > Quorum Call Conference Software > > * Runs on Windows PC. > > * Can connect via VoIP using the international standard SIP protocol. > If VoIP is used no special hardware is needed. > > * Easily allocate new conference IDs on the phone. > > * Join any pre-defined conferences by using the allocated conference > number > > FreeSwitch Conference Features: > > * Software based Conferencing without any hardware requirements. > > * Multiple on-demand or scheduled conferences with entry/exit > announcements > > * Transfers > > * Outbound Calling > > * Configurable Key Lay > > * Bridge to Conference transition > > * Multi Party outbound dialing. > > > -----Original Message----- > From: [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] On Behalf Of Padmaja > Sent: Monday, September 22, 2008 6:15 PM > To: [email protected]; [EMAIL PROTECTED] > Subject: Re: [Sip-implementors] SIP conference server for testing > > Hi, > > Thanks a lot for clearing that doubt. Do you know of any free conference > server supporting the RFC 4579 "Call Control - Conferencing for User > Agents"? I have been searching for a long time but no luck so far. Please > let me know if you know of any. > > Thanks, > Padmaja > > > > Message: 2 > Date: Mon, 15 Sep 2008 15:48:31 +0200 > From: " I?aki Baz Castillo " <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] SIP conference server for testing > Cc: [email protected] > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=UTF-8 > > 2008/9/15, [EMAIL PROTECTED] <[EMAIL PROTECTED]>: > > Asterisk is one open source SIP PBX, which supports conferencing. You > > can try that... > > All SIP Asterisk imlpementation is 100% server side without client > knowledge (in fact, client calls to an Asterisk extension and Asterisk > inserts it into a media conference). The client knows *nothing* (via > SIP I mean) about other participants, and of course it doesn't use SIP > RFC's related to SIP native conference . > > Note that the original question asked for a server supporting > "isfocus" feature (present in RFC 4579 "Call Control - Conferencing > for User Agents"). > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > ________________________________ > "DISCLAIMER: This message is proprietary to Aricent and is intended solely > for the use of the individual to whom it is addressed. It may contain > privileged or confidential information and should not be circulated or used > for any purpose other than for what it is intended. If you have received > this message in error,please notify the originator immediately. If you are > not the intended recipient, you are notified that you are strictly > prohibited from using, copying, altering, or disclosing the contents of this > message. Aricent accepts no responsibility for loss or damage arising from > the use of the information transmitted by this email including damage from > virus." > > > ------------------------------ > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > End of Sip-implementors Digest, Vol 66, Issue 41 > ************************************************ > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
