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From: nermeen <ne...@ci...> - 2011-04-21 15:20:12
|
I also want to reiterate that CTS supports in-band signaling of SPS/PPS as well as out of band signaling of SPS/PPS. In other words CTS always sends SPS/PPS in SDP even when it is also sending them in-band. nermeen On 4/20/11 10:45 PM, "Steve Fry" <sf...@ci...> wrote: > Hi Aravind, > CTS does look at the SPS/PPS and uses these to detect the profile idc, image > height / width, and coding mode (CABAC vs. CAVLC). > If you can please collect the CTS logs from one of the calls and send them to > me, we can take a look at it and try to determine > what the issue is. Also a Wireshark trace of the call startup would be > helpful. > > Thanks > Steve Fry > CTS endpoint team > > > > On 4/20/11 1:32 PM, "Aravind Sethuraman" <ara...@te...> > wrote: > >> Hello all >> >> We are building a TIP aware end point and we want to know if there is any way >> CTS can support in band RTP payload parameter set instead of out of band via >> sprop-parameter-sets. according to the TIP protocol Section 5.2 item #6 >> states that TIP endpoints must indicate support for inband parameter sets. >> >> Currently we are able to setup a call with the cTS 1000 and achieved TIP >> negotiation and have the packet flowing but neither our end point not CTS >> 1000 are able to decode. >> >> any help will be much appreciated >> >> regards >> >> Aravind Sethuraman >> Chief Software Architect >> Teliris, Inc. | 55 Broadway, 14th Floor, New York, NY 10006 | C >> +1.917.355.0119 | O +1.212.490.1065 x1408 | F +1.212.269.2869 | AIM >> aravindsraman >> >> >> ----------------------------------------------------------------------------->> - >> Benefiting from Server Virtualization: Beyond Initial Workload >> Consolidation -- Increasing the use of server virtualization is a top >> priority.Virtualization can reduce costs, simplify management, and improve >> application availability and disaster protection. Learn more about boosting >> the value of server virtualization. http://p.sf.net/sfu/vmware-sfdev2dev >> >> _______________________________________________ >> TIProtocol-askcisco mailing list >> TIP...@li... >> https://lists.sourceforge.net/lists/listinfo/tiprotocol-askcisco > > > ------------------------------------------------------------------------------ > Benefiting from Server Virtualization: Beyond Initial Workload > Consolidation -- Increasing the use of server virtualization is a top > priority.Virtualization can reduce costs, simplify management, and improve > application availability and disaster protection. Learn more about boosting > the value of server virtualization. http://p.sf.net/sfu/vmware-sfdev2dev > > _______________________________________________ > TIProtocol-askcisco mailing list > TIP...@li... > https://lists.sourceforge.net/lists/listinfo/tiprotocol-askcisco |
|
From: David B. (dbenham) <db...@ci...> - 2011-04-21 13:21:32
|
Chenxinxing You should upgrade your CUCM to 8.X http://www.cisco.com/cisco/software/type.html?mdfid=283423434&flowid=213 01 Also upgrade your CTS gear to 1.7.2 http://www.cisco.com/cisco/software/release.html?mdfid=282907611&flowid= 20562&softwareid=280886992&release=1.7.2(4937)&relind=AVAILABLE&rellifec ycle=&reltype=latest > From: Chenxinxing [mailto:che...@hu...] > > What version of CUCM do you have now? > ----- CUCM 6.1.5 > > Cisco will have multiple versions of software, but will > probably conduct most of them with release 1.7.2 > > Huawei coming to SuperOp > ------------------------ > TP1002----single-screen HW Telepresence > TP3006----three-screen HW Telepresence(use small LCD screens) > SMC----Service Management Center(like a call manager) > MCU----Multipoint Control Unit > HW Ethernet Switch > Wireshark, FTP, Secure-CRT, etc > |
|
From: Steve F. <sf...@ci...> - 2011-04-21 05:45:52
|
Hi Aravind, CTS does look at the SPS/PPS and uses these to detect the profile idc, image height / width, and coding mode (CABAC vs. CAVLC). If you can please collect the CTS logs from one of the calls and send them to me, we can take a look at it and try to determine what the issue is. Also a Wireshark trace of the call startup would be helpful. Thanks Steve Fry CTS endpoint team On 4/20/11 1:32 PM, "Aravind Sethuraman" <ara...@te...> wrote: > Hello all > > We are building a TIP aware end point and we want to know if there is any way > CTS can support in band RTP payload parameter set instead of out of band via > sprop-parameter-sets. according to the TIP protocol Section 5.2 item #6 states > that TIP endpoints must indicate support for inband parameter sets. > > Currently we are able to setup a call with the cTS 1000 and achieved TIP > negotiation and have the packet flowing but neither our end point not CTS 1000 > are able to decode. > > any help will be much appreciated > > regards > > Aravind Sethuraman > Chief Software Architect > Teliris, Inc. | 55 Broadway, 14th Floor, New York, NY 10006 | C > +1.917.355.0119 | O +1.212.490.1065 x1408 | F +1.212.269.2869 | AIM > aravindsraman > > > ------------------------------------------------------------------------------ > Benefiting from Server Virtualization: Beyond Initial Workload > Consolidation -- Increasing the use of server virtualization is a top > priority.Virtualization can reduce costs, simplify management, and improve > application availability and disaster protection. Learn more about boosting > the value of server virtualization. http://p.sf.net/sfu/vmware-sfdev2dev > > _______________________________________________ > TIProtocol-askcisco mailing list > TIP...@li... > https://lists.sourceforge.net/lists/listinfo/tiprotocol-askcisco |
|
From: Aravind S. <ara...@te...> - 2011-04-21 03:22:05
|
Hello all We are building a TIP aware end point and we want to know if there is any way CTS can support in band RTP payload parameter set instead of out of band via sprop-parameter-sets. according to the TIP protocol Section 5.2 item #6 states that TIP endpoints must indicate support for inband parameter sets. Currently we are able to setup a call with the cTS 1000 and achieved TIP negotiation and have the packet flowing but neither our end point not CTS 1000 are able to decode. any help will be much appreciated regards Aravind Sethuraman Chief Software Architect Teliris, Inc. | 55 Broadway, 14th Floor, New York, NY 10006 | C +1.917.355.0119 | O +1.212.490.1065 x1408 | F +1.212.269.2869 | AIM aravindsraman |
|
From: nermeen <ne...@ci...> - 2011-04-20 23:10:03
|
Hi Aravind, 1.7 CTS will include SPS and PPS in its SIP signaling even when it has negotiated and is actually sending them in-band. nermeen On 4/20/11 3:26 PM, "Aravind Sethuraman" <ara...@te...> wrote: > Nermeen, > We have configured the default single system as per the tip library and it > does > addVideoMediaOption(system, CTipVideoMediaOption::INBAND_PARAM_SETS, > CTipMediaOption::OPTION_SUPPORTED_BOTH) > in the tip_profile module. > > does this mean is adhering to sending inband param sets although it has > signalled out-of-band in the 183? > > thanks > aravind > > On 04/20/2011 05:36 PM, nermeen wrote: >> Re: [TIProtocol-askcisco] FW: Teliris & TIP Protocol Start-up Issues >> Aravind, >> >> Can you let us know what has been signaled in the TIP MediaOpts packet in >> terms of inband support? What has your device transmitted to the CTS and what >> has it received from the CTS? >> >> If the remote device has indicated support for the reception of in band >> signaling in its MediaOpts then the CTS will transmit inband signaling >> otherwise it would not. >> >> nermeen >> >> >> On 4/20/11 1:22 PM, "David Benham (dbenham)" <db...@ci...> wrote: >> >> >>> Aravind >>> Go ahead and copy the mail list ( tip...@li... >>> ), which I have done here, so the rest of the team at Cisco sees the >>> questions/issues, especially since I am off-site the rest of this week. >>> >>> >>> >>> From: Aravind Sethuraman [mailto:ara...@te...] >>> Sent: Wednesday, April 20, 2011 1:12 PM >>> To: David Benham (dbenham); Girish Kondappa >>> Subject: Re: FW: Teliris & TIP Protocol Start-up Issues >>> Importance: High >>> >>> Hi David >>> >>> Thank you for getting back to us. The issue was the bandwidth in CUCM - >>> changing it enabled us to be in a call. However, we are seeing CTS is >>> sending only out of band parameter sets but the TIP Protocol mandates that >>> all TIP must support in band >>> Section 5.2 point #6 >>> Is there another configuration issue? >>> >>> much obliged >>> Aravind >>> >>> >>> On 04/20/2011 03:26 PM, David Benham (dbenham) wrote: >>> Aravind >>> >>> Two issues popped up in our quick review >>> >>> 1) Not enough bandwidth in the offer for Video; 320K. >>> Venturing a guess, have you configured the CUCM¹s default region bandwidth? >>> >>> If not, do the following after logging in to the CUCM >>> System-->Region, click ³find², then the default region will show. Click >>> ³default². >>> Go to ³Modify Relationship to other Regions, and select ³Default². >>> Under Max Video Call Bit Rate, enter ³32000², and select the ³kpbs² >>> radio button. >>> Click ³Save², then ³Reset². >>> >>> 2) The ACK SIP to our SIP 200 OK only indicates G.711, so be sure you >>> offer/ack AAC-LD first or only >>> >>> m=audio 53008 RTP/AVP 0^M >>> b=TIAS:64000^M >>> a=rtpmap:0 PCMU/8000^M >>> a=ptime:20^M >>> >>> >>> >>> >>> From: Aravind Sethuraman [mailto:ara...@te...] >>> Sent: Tuesday, April 19, 2011 2:01 PM >>> To: David Benham (dbenham) >>> Cc: ste...@te...; tip...@so... >>> Subject: Re: Teliris & TIP Protocol Start-up Issues >>> Importance: High >>> >>> David, >>> >>> Thanks for getting back.. >>> As per your request attached are the invite.txt and the cts log files >>> this time, the behaviour was CTS reports Configuration mismatch.... >>> >>> much obliged >>> Aravind >>> >>> >>> >>> On 04/19/2011 04:12 PM, David Benham (dbenham) wrote: >>> Aravind, >>> >>> Send us a full copy of your initial SIP INVITE containing the SDP. >>> Also send a copy of the CTS¹ logs just in case. >>> >>> >>> From: Aravind Sethuraman [mailto:ara...@te...] >>> Sent: Tuesday, April 19, 2011 9:31 AM >>> To: ste...@te... >>> Cc: David Benham (dbenham) >>> Subject: Re: Teliris & TIP Protocol License Agreement >>> >>> Hello David >>> Firstly i would like to thank you for offering your help. >>> What we are building is a TIP user agent plugin which acts as a B2BUA >>> talking TIP on one side and SIP/H264SVC on the other side. Our ultimate goal >>> is to build this to interop with Cisco CTS family of products, 1000-3000. We >>> are currently testing with a CUCM version 8.5 and CTS 1000 version 1.7 and >>> we have chosen to adopt TIP profile version 6. >>> >>> The TIPUA does support the following >>> >>> 1. SIP signaling >>> 2. Video >>> 3. >>>> 1. supports H264 >>>> 2. supports b=TIAS >>>> 3. supports RTP format 112 and packetization mode >>>> 4. supports the profile-id - 4d0028 >>>> 5. >>> 4. Audio >>> 5. >>>> 1. AAC-LD (we are faking it currently but @ signal level we are doing it to >>>> make CTS happy. We will eventually have this up and running) >>>> 2. G711 >>>> 3. >>> 6. KeyPress >>> 7. >>>> 1. RFC2833 ONLY >>>> 2. >>> 8. TIP Negotiation -> We are having the TIP negotiation module which i will >>> define further below... >>> 9. >>> What we are seeing is as follows:- >>> >>> 1. Our TIPUA sends invite to CTS (both are registered to CUCM) >>> 2. CTS sends its SDP in 183 >>> 3. upon receipt of 183, we start the TIP negotiation -> note, We are >>> assuming the RTCP port available is RTP+1 (in the SDP) >>> 4. >>>> 1. We are using the libTIP version 1.3 library. >>>> 2. >>> 5. eventually CTS sends a mid-call re-invite and downgrades to audio only >>> call. >>> 6. We have not been able to capture ANY RTCP/TIP packets from the CTS >>> leading to eventual failure of TIP and therefore AUDIO only call. >>> 7. >>> I am at my wits' end. Are we doing anything glaringly wrong? does CTS 1.7 >>> NOT support TIP ?(highly unlikely) >>> >>> I am also available for any phone call if you feel it is necessary to have @ >>> 1917-355-0119 >>> >>> much obliged >>> Aravind Sethuraman >>> >>> >>> >>> >>> >>> >>> >>> >>> On 04/13/2011 05:42 PM, Steven Gage wrote: >>> David, >>> >>> Aravind is our Chief Software Architect and the technical lead on our >>> Interop efforts. >>> >>> Aravind, >>> >>> David has been extremely helpful in providing direction on TIP >>> Interoperability. >>> >>> Please document our assumptions specifically re sip and tip end point >>> connectivity, minimum bandwidth requirements for video, and audio codec >>> requirements. >>> >>> At David's suggestion we will be attending superop but would like to get the >>> basics working in our lab. >>> >>> Regards, >>> Steven Gage >>> Teliris >>> 55 Broadway | NY NY 10006 >>> O+1.212.490.1065 x1400 >>> F+1.212.202.5432 >>> M+1.917.952.2212 >>> >>> ---------------------------------------------------------------------------- >>> >>> This message is a PRIVATE communication. This message and all attachments >>> are a private communication are confidential or >>> protected by privilege. If you are not the intended recipient, you are >>> hereby notified that any disclosure, copying, distribution or use of the >>> information contained in or attached to this message is strictly prohibited. >>> Please notify the sender of the delivery error by replying to this message, >>> and then delete it from your system. Thank you. >>> >>> >>> >>> >>> >>> >>> From: "David Benham (dbenham)" <db...@ci...> >>> <mailto:db...@ci...> >>> >>> Date: Wed, 13 Apr 2011 12:58:02 -0700 >>> >>> To: Steven Gage<ste...@te...> <mailto:ste...@te...> >>> >>> Subject: RE: Teliris & TIP Protocol License Agreement >>> >>> >>> Hi Steven >>> >>> The TIP protocol specification does recommend (³SHOULD²), but does not >>> mandate, that endpoints default back to a p2p video and audio call if TIP is >>> not negotiated. The CTS 1000 you have will default back to an audio only >>> (G.7xx) call, in such a case. >>> >>> Also, when you do attempt to negotiate TIP with that CTS 1000, you need to >>> indicate AAC-LD in the SDP audio line (per sect 3.1 in the Cisco TIP >>> Endpoint Implementation Profile). Otherwise, the CTS endpoint will assume >>> it is not a TIP session and resort to an audio-only call. >>> >>> Let us know if this helps or can answer more questions. >>> >>> >>> >>> From: Steven Gage [mailto:ste...@te...] >>> Sent: Wednesday, April 06, 2011 6:56 PM >>> To: David Benham (dbenham) >>> Subject: Re: Teliris & TIP Protocol License Agreement >>> Importance: High >>> >>> David, >>> >>> >>> >>> I posted our issue, here is a synopsis: >>> >>> >>> >>> We are trying to setup call between a CTS 1000 and SIP Video Endpoint >>> (Mirial ver 7.0.42). >>> >>> >>> >>> The Mirial client is a pure SIP Video Endpoint capable of supporting H264 >>> AVC and G711. >>> >>> >>> >>> We have registered it to the CUCM (version 8.5) as a 3rd party Advanced SIP >>> Endpoint with Digest authentication >>> >>> >>> >>> We have a CTS 1000 registered to same CUCM >>> >>> >>> >>> While trying to setup call between the CTS 1000 and SIP Video Endpoint >>> (Mirial ver 7.0.42), we are not able to setup a video call. We see in the >>> SIP Packet Transfer, that CUCM re-writes (there are 2 >>> >>> SIP Invites sent, but that gets fairly in depth SIP wise) the invite as >>> audio only invite. >>> >>> >>> >>> Basically we need to figure out what is the configuration (do we have to >>> add a SIP Trunk etc) on the CUCM to make it recognize/support video calls >>> with a pure SIP Video Endpoint. >>> >>> >>> >>> >>> >>> Thanks, >>> >>> >>> >>> >>> >>> Steve >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Apr 2, 2011, at 10:36 AM, David Benham (dbenham) wrote: >>> >>> >>> >>> >>> Steven >>> >>> Considering it is only 6 weeks away, I strongly suggest attending IMTC¹s >>> SuperOp event, where Cisco and Polycom (at least) will have gear set up to >>> test with each other¹s TIP products. Non-members are invited to this >>> SuperOp, but do note that if you join IMTC, you can also partake in the TIP >>> Activity Group¹s stewardship of TIP as well as other membership benefits. >>> >>> >>> >>> Here are links and email addresses for further query. >>> >>> >>> >>> IMTC page for TIP Developers >>> >>> http://www.imtc.org/tip/ >>> >>> >>> >>> IMTC page for TIP Activity Group >>> >>> http://www.imtc.org/activity_groups/tip.asp >>> >>> >>> >>> Places to post questions tip...@im... >>> >>> or at the TIP open source project >>> >>> http://sourceforge.net/projects/tiprotocol/support >>> >>> >>> >>> IMTC page for SuperOp event (top item on page) >>> >>> http://www.imtc.org/events/ >>> >>> >>> >>> Cisco page for TIP Developers >>> >>> http://www.cisco.com/go/tip >>> >>> >>> >>> >>> >>> >>> >>> >>> ---------------------------------------------------------------------------- >>> -- >>> Benefiting from Server Virtualization: Beyond Initial Workload >>> Consolidation -- Increasing the use of server virtualization is a top >>> priority.Virtualization can reduce costs, simplify management, and improve >>> application availability and disaster protection. Learn more about boosting >>> the value of server virtualization. http://p.sf.net/sfu/vmware-sfdev2dev >>> >>> >>> _______________________________________________ >>> TIProtocol-askcisco mailing list >>> TIP...@li... >>> https://lists.sourceforge.net/lists/listinfo/tiprotocol-askcisco >>> >>> >>> >>> |
|
From: Aravind S. <ara...@te...> - 2011-04-20 22:27:07
|
Nermeen,
We have configured the default single system as per the tip library and
it does
addVideoMediaOption(system, CTipVideoMediaOption::INBAND_PARAM_SETS,
CTipMediaOption::OPTION_SUPPORTED_BOTH)
in the tip_profile module.
does this mean is adhering to sending inband param sets although it has
signalled out-of-band in the 183?
thanks
aravind
On 04/20/2011 05:36 PM, nermeen wrote:
> Aravind,
>
> Can you let us know what has been signaled in the TIP MediaOpts packet
> in terms of inband support? What has your device transmitted to the
> CTS and what has it received from the CTS?
>
> If the remote device has indicated support for the reception of in
> band signaling in its MediaOpts then the CTS will transmit inband
> signaling otherwise it would not.
>
> nermeen
>
>
> On 4/20/11 1:22 PM, "David Benham (dbenham)" <db...@ci...> wrote:
>
> Aravind
> Go ahead and copy the mail list (
> tip...@li... ), which I have done
> here, so the rest of the team at Cisco sees the questions/issues,
> especially since I am off-site the rest of this week.
>
>
>
> *From:* Aravind Sethuraman [mailto:ara...@te...]
> *Sent:* Wednesday, April 20, 2011 1:12 PM
> *To:* David Benham (dbenham); Girish Kondappa
> *Subject:* Re: FW: Teliris & TIP Protocol Start-up Issues
> *Importance:* High
>
> Hi David
>
> Thank you for getting back to us. The issue was the bandwidth in
> CUCM - changing it enabled us to be in a call. However, we are
> seeing CTS is sending only out of band parameter sets but the TIP
> Protocol mandates that all TIP must support in band
> Section 5.2 point #6
> Is there another configuration issue?
>
> much obliged
> Aravind
>
>
> On 04/20/2011 03:26 PM, David Benham (dbenham) wrote:
> Aravind
>
> Two issues popped up in our quick review ...
>
> 1) Not enough bandwidth in the offer for Video; 320K.
> Venturing a guess, have you configured the CUCM's default region
> bandwidth?
>
> If not, do the following after logging in to the CUCM
> System-->Region, click "find", then the default region will
> show. Click "default".
> Go to "Modify Relationship to other Regions, and select "Default".
> Under Max Video Call Bit Rate, enter "32000", and select the
> "kpbs" radio button.
> Click "Save", then "Reset".
>
> 2) The ACK SIP to our SIP 200 OK only indicates G.711, so be
> sure you offer/ack AAC-LD first or only
>
> m=audio 53008 RTP/AVP 0^M
> b=TIAS:64000^M
> a=rtpmap:0 PCMU/8000^M
> a=ptime:20^M
>
>
>
>
> *From:* Aravind Sethuraman [mailto:ara...@te...]
> *Sent:* Tuesday, April 19, 2011 2:01 PM
> *To:* David Benham (dbenham)
> *Cc:* ste...@te...; tip...@so...
> *Subject:* Re: Teliris & TIP Protocol Start-up Issues
> *Importance:* High
>
> David,
>
> Thanks for getting back..
> As per your request attached are the invite.txt and the cts log files
> this time, the behaviour was CTS reports Configuration mismatch....
>
> much obliged
> Aravind
>
>
>
> On 04/19/2011 04:12 PM, David Benham (dbenham) wrote:
> Aravind,
>
> Send us a full copy of your initial SIP INVITE containing the SDP.
> Also send a copy of the CTS' logs just in case.
>
>
> *From:* Aravind Sethuraman [mailto:ara...@te...]
> *Sent:* Tuesday, April 19, 2011 9:31 AM
> *To:* ste...@te...
> *Cc:* David Benham (dbenham)
> *Subject:* Re: Teliris & TIP Protocol License Agreement
>
> Hello David
> Firstly i would like to thank you for offering your help.
> What we are building is a TIP user agent plugin which acts as a
> B2BUA talking TIP on one side and SIP/H264SVC on the other side.
> Our ultimate goal is to build this to interop with Cisco CTS
> family of products, 1000-3000. We are currently testing with a
> CUCM version 8.5 and CTS 1000 version 1.7 and we have chosen to
> adopt TIP profile version 6.
>
> The TIPUA does support the following
>
> 1. SIP signaling
> 2. Video
> 1. supports H264
> 2. supports b=TIAS
> 3. supports RTP format 112 and packetization mode
> 4. supports the profile-id - 4d0028
> 3. Audio
> 1. AAC-LD (we are faking it currently but @ signal level
> we are doing it to make CTS happy. We will eventually
> have this up and running)
> 2. G711
> 4. KeyPress
> 1. RFC2833 ONLY
> 5. TIP Negotiation -> We are having the TIP negotiation module
> which i will define further below...
>
> What we are seeing is as follows:-
>
> 1. Our TIPUA sends invite to CTS (both are registered to CUCM)
> 2. CTS sends its SDP in 183
> 3. upon receipt of 183, we start the TIP negotiation -> note,
> We are assuming the RTCP port available is RTP+1 (in the SDP)
> 1. We are using the libTIP version 1.3 library.
> 4. eventually CTS sends a mid-call re-invite and downgrades to
> audio only call.
> 5. We have not been able to capture ANY RTCP/TIP packets from
> the CTS leading to eventual failure of TIP and therefore
> AUDIO only call.
>
> I am at my wits' end. Are we doing anything glaringly wrong? does
> CTS 1.7 NOT support TIP ?(highly unlikely)
>
> I am also available for any phone call if you feel it is necessary
> to have @ 1917-355-0119
>
> much obliged
> Aravind Sethuraman
>
>
>
>
>
>
>
>
> On 04/13/2011 05:42 PM, Steven Gage wrote:
> David,
>
> Aravind is our Chief Software Architect and the technical lead on
> our Interop efforts.
>
> Aravind,
>
> David has been extremely helpful in providing direction on TIP
> Interoperability.
>
> Please document our assumptions specifically re sip and tip end
> point connectivity, minimum bandwidth requirements for video, and
> audio codec requirements.
>
> At David's suggestion we will be attending superop but would like
> to get the basics working in our lab.
>
> Regards,
> Steven Gage
> Teliris
> 55 Broadway | NY NY 10006
> O+1.212.490.1065 x1400
> F+1.212.202.5432
> M+1.917.952.2212
>
> ----------------------------------------------------------------------------
>
> This message is a PRIVATE communication. This message and all
> attachments are a private communication are confidential or
> protected by privilege. If you are not the intended recipient, you are
> hereby notified that any disclosure, copying, distribution or use
> of the information contained in or attached to this message is
> strictly prohibited. Please notify the sender of the delivery
> error by replying to this message, and then delete it from your
> system. Thank you.
>
> ------------------------------------------------------------------------
>
> *From: *"David Benham (dbenham)" <db...@ci...>
> <mailto:db...@ci...>
>
> *Date: *Wed, 13 Apr 2011 12:58:02 -0700
>
> *To: *Steven Gage<ste...@te...>
> <mailto:ste...@te...>
>
> *Subject: *RE: Teliris & TIP Protocol License Agreement
>
>
> Hi Steven
>
> The TIP protocol specification does recommend ("SHOULD"), but does
> not mandate, that endpoints default back to a p2p video and audio
> call if TIP is not negotiated. The CTS 1000 you have will default
> back to an audio only (G.7xx) call, in such a case.
>
> Also, when you do attempt to negotiate TIP with that CTS 1000, you
> need to indicate AAC-LD in the SDP audio line (per sect 3.1 in the
> Cisco TIP Endpoint Implementation Profile). Otherwise, the CTS
> endpoint will assume it is not a TIP session and resort to an
> audio-only call.
>
> Let us know if this helps or can answer more questions.
>
>
>
> *From:* Steven Gage [mailto:ste...@te...]
> *Sent:* Wednesday, April 06, 2011 6:56 PM
> *To:* David Benham (dbenham)
> *Subject:* Re: Teliris & TIP Protocol License Agreement
> *Importance:* High
>
> David,
>
>
>
> I posted our issue, here is a synopsis:
>
>
>
> We are trying to setup call between a CTS 1000 and SIP Video
> Endpoint (Mirial ver 7.0.42).
>
>
>
> The Mirial client is a pure SIP Video Endpoint capable of
> supporting H264 AVC and G711.
>
>
>
> We have registered it to the CUCM (version 8.5) as a 3rd party
> Advanced SIP Endpoint with Digest authentication
>
>
>
> We have a CTS 1000 registered to same CUCM
>
>
>
> While trying to setup call between the CTS 1000 and SIP Video
> Endpoint (Mirial ver 7.0.42), we are not able to setup a video
> call. We see in the SIP Packet Transfer, that CUCM re-writes
> (there are 2
>
> SIP Invites sent, but that gets fairly in depth SIP wise) the
> invite as audio only invite.
>
>
>
> Basically we need to figure out what is the configuration (do we
> have to add a SIP Trunk etc) on the CUCM to make it
> recognize/support video calls with a pure SIP Video Endpoint.
>
>
>
>
>
> Thanks,
>
>
>
>
>
> Steve
>
>
>
>
>
>
>
>
>
>
>
> On Apr 2, 2011, at 10:36 AM, David Benham (dbenham) wrote:
>
>
>
>
> Steven
>
> Considering it is only 6 weeks away, I strongly suggest attending
> IMTC's SuperOp event, where Cisco and Polycom (at least) will have
> gear set up to test with each other's TIP products. Non-members
> are invited to this SuperOp, but do note that if you join IMTC,
> you can also partake in the TIP Activity Group's stewardship of
> TIP as well as other membership benefits.
>
>
>
> Here are links and email addresses for further query.
>
>
>
> IMTC page for TIP Developers
>
> http://www.imtc.org/tip/
>
>
>
> IMTC page for TIP Activity Group
>
> http://www.imtc.org/activity_groups/tip.asp
>
>
>
> Places to post questions tip...@im...
>
> or at the TIP open source project
>
> http://sourceforge.net/projects/tiprotocol/support
>
>
>
> IMTC page for SuperOp event (top item on page)
>
> http://www.imtc.org/events/
>
>
>
> Cisco page for TIP Developers
>
> http://www.cisco.com/go/tip
>
>
>
>
>
>
> ------------------------------------------------------------------------
> ------------------------------------------------------------------------------
> Benefiting from Server Virtualization: Beyond Initial Workload
> Consolidation -- Increasing the use of server virtualization is a top
> priority.Virtualization can reduce costs, simplify management, and
> improve
> application availability and disaster protection. Learn more about
> boosting
> the value of server virtualization.
> http://p.sf.net/sfu/vmware-sfdev2dev
> ------------------------------------------------------------------------
> _______________________________________________
> TIProtocol-askcisco mailing list
> TIP...@li...
> https://lists.sourceforge.net/lists/listinfo/tiprotocol-askcisco
>
|
|
From: nermeen <ne...@ci...> - 2011-04-20 21:36:03
|
Aravind, Can you let us know what has been signaled in the TIP MediaOpts packet in terms of inband support? What has your device transmitted to the CTS and what has it received from the CTS? If the remote device has indicated support for the reception of in band signaling in its MediaOpts then the CTS will transmit inband signaling otherwise it would not. nermeen On 4/20/11 1:22 PM, "David Benham (dbenham)" <db...@ci...> wrote: > Aravind > Go ahead and copy the mail list ( tip...@li... ), > which I have done here, so the rest of the team at Cisco sees the > questions/issues, especially since I am off-site the rest of this week. > > > > From: Aravind Sethuraman [mailto:ara...@te...] > Sent: Wednesday, April 20, 2011 1:12 PM > To: David Benham (dbenham); Girish Kondappa > Subject: Re: FW: Teliris & TIP Protocol Start-up Issues > Importance: High > > Hi David > > Thank you for getting back to us. The issue was the bandwidth in CUCM - > changing it enabled us to be in a call. However, we are seeing CTS is sending > only out of band parameter sets but the TIP Protocol mandates that all TIP > must support in band > Section 5.2 point #6 > Is there another configuration issue? > > much obliged > Aravind > > > On 04/20/2011 03:26 PM, David Benham (dbenham) wrote: > Aravind > > Two issues popped up in our quick review > > 1) Not enough bandwidth in the offer for Video; 320K. > Venturing a guess, have you configured the CUCM¹s default region bandwidth? > > If not, do the following after logging in to the CUCM > System-->Region, click ³find², then the default region will show. Click > ³default². > Go to ³Modify Relationship to other Regions, and select ³Default². > Under Max Video Call Bit Rate, enter ³32000², and select the ³kpbs² radio > button. > Click ³Save², then ³Reset². > > 2) The ACK SIP to our SIP 200 OK only indicates G.711, so be sure you > offer/ack AAC-LD first or only > > m=audio 53008 RTP/AVP 0^M > b=TIAS:64000^M > a=rtpmap:0 PCMU/8000^M > a=ptime:20^M > > > > > From: Aravind Sethuraman [mailto:ara...@te...] > Sent: Tuesday, April 19, 2011 2:01 PM > To: David Benham (dbenham) > Cc: ste...@te...; tip...@so... > Subject: Re: Teliris & TIP Protocol Start-up Issues > Importance: High > > David, > > Thanks for getting back.. > As per your request attached are the invite.txt and the cts log files > this time, the behaviour was CTS reports Configuration mismatch.... > > much obliged > Aravind > > > > On 04/19/2011 04:12 PM, David Benham (dbenham) wrote: > Aravind, > > Send us a full copy of your initial SIP INVITE containing the SDP. > Also send a copy of the CTS¹ logs just in case. > > > From: Aravind Sethuraman [mailto:ara...@te...] > Sent: Tuesday, April 19, 2011 9:31 AM > To: ste...@te... > Cc: David Benham (dbenham) > Subject: Re: Teliris & TIP Protocol License Agreement > > Hello David > Firstly i would like to thank you for offering your help. > What we are building is a TIP user agent plugin which acts as a B2BUA talking > TIP on one side and SIP/H264SVC on the other side. Our ultimate goal is to > build this to interop with Cisco CTS family of products, 1000-3000. We are > currently testing with a CUCM version 8.5 and CTS 1000 version 1.7 and we have > chosen to adopt TIP profile version 6. > > The TIPUA does support the following > 1. SIP signaling > 2. Video >> 1. supports H264 >> 2. supports b=TIAS >> 3. supports RTP format 112 and packetization mode >> 4. supports the profile-id - 4d0028 > 3. Audio >> 1. AAC-LD (we are faking it currently but @ signal level we are doing it to >> make CTS happy. We will eventually have this up and running) >> 2. G711 > 4. KeyPress >> 1. RFC2833 ONLY > 5. TIP Negotiation -> We are having the TIP negotiation module which i will > define further below... > What we are seeing is as follows:- > 1. Our TIPUA sends invite to CTS (both are registered to CUCM) > 2. CTS sends its SDP in 183 > 3. upon receipt of 183, we start the TIP negotiation -> note, We are assuming > the RTCP port available is RTP+1 (in the SDP) >> 1. We are using the libTIP version 1.3 library. > 4. eventually CTS sends a mid-call re-invite and downgrades to audio only > call. > 5. We have not been able to capture ANY RTCP/TIP packets from the CTS leading > to eventual failure of TIP and therefore AUDIO only call. > I am at my wits' end. Are we doing anything glaringly wrong? does CTS 1.7 NOT > support TIP ?(highly unlikely) > > I am also available for any phone call if you feel it is necessary to have @ > 1917-355-0119 > > much obliged > Aravind Sethuraman > > > > > > > > > On 04/13/2011 05:42 PM, Steven Gage wrote: > David, > > Aravind is our Chief Software Architect and the technical lead on our Interop > efforts. > > Aravind, > > David has been extremely helpful in providing direction on TIP > Interoperability. > > Please document our assumptions specifically re sip and tip end point > connectivity, minimum bandwidth requirements for video, and audio codec > requirements. > > At David's suggestion we will be attending superop but would like to get the > basics working in our lab. > > Regards, > Steven Gage > Teliris > 55 Broadway | NY NY 10006 > O+1.212.490.1065 x1400 > F+1.212.202.5432 > M+1.917.952.2212 > > ---------------------------------------------------------------------------- > > This message is a PRIVATE communication. This message and all attachments are > a private communication are confidential or > protected by privilege. If you are not the intended recipient, you are > hereby notified that any disclosure, copying, distribution or use of the > information contained in or attached to this message is strictly prohibited. > Please notify the sender of the delivery error by replying to this message, > and then delete it from your system. Thank you. > > > From: "David Benham (dbenham)" <db...@ci...> <mailto:db...@ci...> > > Date: Wed, 13 Apr 2011 12:58:02 -0700 > > To: Steven Gage<ste...@te...> <mailto:ste...@te...> > > Subject: RE: Teliris & TIP Protocol License Agreement > > > Hi Steven > > The TIP protocol specification does recommend (³SHOULD²), but does not > mandate, that endpoints default back to a p2p video and audio call if TIP is > not negotiated. The CTS 1000 you have will default back to an audio only > (G.7xx) call, in such a case. > > Also, when you do attempt to negotiate TIP with that CTS 1000, you need to > indicate AAC-LD in the SDP audio line (per sect 3.1 in the Cisco TIP Endpoint > Implementation Profile). Otherwise, the CTS endpoint will assume it is not a > TIP session and resort to an audio-only call. > > Let us know if this helps or can answer more questions. > > > > From: Steven Gage [mailto:ste...@te...] > Sent: Wednesday, April 06, 2011 6:56 PM > To: David Benham (dbenham) > Subject: Re: Teliris & TIP Protocol License Agreement > Importance: High > > David, > > > > I posted our issue, here is a synopsis: > > > > We are trying to setup call between a CTS 1000 and SIP Video Endpoint (Mirial > ver 7.0.42). > > > > The Mirial client is a pure SIP Video Endpoint capable of supporting H264 AVC > and G711. > > > > We have registered it to the CUCM (version 8.5) as a 3rd party Advanced SIP > Endpoint with Digest authentication > > > > We have a CTS 1000 registered to same CUCM > > > > While trying to setup call between the CTS 1000 and SIP Video Endpoint (Mirial > ver 7.0.42), we are not able to setup a video call. We see in the SIP Packet > Transfer, that CUCM re-writes (there are 2 > > SIP Invites sent, but that gets fairly in depth SIP wise) the invite as audio > only invite. > > > > Basically we need to figure out what is the configuration (do we have to add a > SIP Trunk etc) on the CUCM to make it recognize/support video calls with a > pure SIP Video Endpoint. > > > > > > Thanks, > > > > > > Steve > > > > > > > > > > > > On Apr 2, 2011, at 10:36 AM, David Benham (dbenham) wrote: > > > > > Steven > > Considering it is only 6 weeks away, I strongly suggest attending IMTC¹s > SuperOp event, where Cisco and Polycom (at least) will have gear set up to > test with each other¹s TIP products. Non-members are invited to this > SuperOp, but do note that if you join IMTC, you can also partake in the TIP > Activity Group¹s stewardship of TIP as well as other membership benefits. > > > > Here are links and email addresses for further query. > > > > IMTC page for TIP Developers > > http://www.imtc.org/tip/ > > > > IMTC page for TIP Activity Group > > http://www.imtc.org/activity_groups/tip.asp > > > > Places to post questions tip...@im... > > or at the TIP open source project > > http://sourceforge.net/projects/tiprotocol/support > > > > IMTC page for SuperOp event (top item on page) > > http://www.imtc.org/events/ > > > > Cisco page for TIP Developers > > http://www.cisco.com/go/tip > > > > > > > > ------------------------------------------------------------------------------ > Benefiting from Server Virtualization: Beyond Initial Workload > Consolidation -- Increasing the use of server virtualization is a top > priority.Virtualization can reduce costs, simplify management, and improve > application availability and disaster protection. Learn more about boosting > the value of server virtualization. http://p.sf.net/sfu/vmware-sfdev2dev > > _______________________________________________ > TIProtocol-askcisco mailing list > TIP...@li... > https://lists.sourceforge.net/lists/listinfo/tiprotocol-askcisco |
|
From: David B. (dbenham) <db...@ci...> - 2011-04-20 20:22:37
|
Aravind
Go ahead and copy the mail list (
tip...@li... ), which I have done here, so
the rest of the team at Cisco sees the questions/issues, especially
since I am off-site the rest of this week.
From: Aravind Sethuraman [mailto:ara...@te...]
Sent: Wednesday, April 20, 2011 1:12 PM
To: David Benham (dbenham); Girish Kondappa
Subject: Re: FW: Teliris & TIP Protocol Start-up Issues
Importance: High
Hi David
Thank you for getting back to us. The issue was the bandwidth in CUCM -
changing it enabled us to be in a call. However, we are seeing CTS is
sending only out of band parameter sets but the TIP Protocol mandates
that all TIP must support in band
Section 5.2 point #6
Is there another configuration issue?
much obliged
Aravind
On 04/20/2011 03:26 PM, David Benham (dbenham) wrote:
Aravind
Two issues popped up in our quick review ...
1) Not enough bandwidth in the offer for Video; 320K.
Venturing a guess, have you configured the CUCM's default region
bandwidth?
If not, do the following after logging in to the CUCM
System-->Region, click "find", then the default region will show.
Click "default".
Go to "Modify Relationship to other Regions, and select "Default".
Under Max Video Call Bit Rate, enter "32000", and select the "kpbs"
radio button.
Click "Save", then "Reset".
2) The ACK SIP to our SIP 200 OK only indicates G.711, so be sure you
offer/ack AAC-LD first or only
m=audio 53008 RTP/AVP 0^M
b=TIAS:64000^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M
From: Aravind Sethuraman [mailto:ara...@te...]
Sent: Tuesday, April 19, 2011 2:01 PM
To: David Benham (dbenham)
Cc: ste...@te...; tip...@so...
Subject: Re: Teliris & TIP Protocol Start-up Issues
Importance: High
David,
Thanks for getting back..
As per your request attached are the invite.txt and the cts log files
this time, the behaviour was CTS reports Configuration mismatch....
much obliged
Aravind
On 04/19/2011 04:12 PM, David Benham (dbenham) wrote:
Aravind,
Send us a full copy of your initial SIP INVITE containing the SDP.
Also send a copy of the CTS' logs just in case.
From: Aravind Sethuraman [mailto:ara...@te...]
Sent: Tuesday, April 19, 2011 9:31 AM
To: ste...@te...
Cc: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement
Hello David
Firstly i would like to thank you for offering your help.
What we are building is a TIP user agent plugin which acts as a B2BUA
talking TIP on one side and SIP/H264SVC on the other side. Our ultimate
goal is to build this to interop with Cisco CTS family of products,
1000-3000. We are currently testing with a CUCM version 8.5 and CTS 1000
version 1.7 and we have chosen to adopt TIP profile version 6.
The TIPUA does support the following
1. SIP signaling
2. Video
1. supports H264
2. supports b=TIAS
3. supports RTP format 112 and packetization mode
4. supports the profile-id - 4d0028
3. Audio
1. AAC-LD (we are faking it currently but @ signal level we
are doing it to make CTS happy. We will eventually have this up and
running)
2. G711
4. KeyPress
1. RFC2833 ONLY
5. TIP Negotiation -> We are having the TIP negotiation module
which i will define further below...
What we are seeing is as follows:-
1. Our TIPUA sends invite to CTS (both are registered to CUCM)
2. CTS sends its SDP in 183
3. upon receipt of 183, we start the TIP negotiation -> note, We
are assuming the RTCP port available is RTP+1 (in the SDP)
1. We are using the libTIP version 1.3 library.
4. eventually CTS sends a mid-call re-invite and downgrades to
audio only call.
5. We have not been able to capture ANY RTCP/TIP packets from the
CTS leading to eventual failure of TIP and therefore AUDIO only call.
I am at my wits' end. Are we doing anything glaringly wrong? does CTS
1.7 NOT support TIP ?(highly unlikely)
I am also available for any phone call if you feel it is necessary to
have @ 1917-355-0119
much obliged
Aravind Sethuraman
On 04/13/2011 05:42 PM, Steven Gage wrote:
David,
Aravind is our Chief Software Architect and the technical lead on our
Interop efforts.
Aravind,
David has been extremely helpful in providing direction on TIP
Interoperability.
Please document our assumptions specifically re sip and tip end point
connectivity, minimum bandwidth requirements for video, and audio codec
requirements.
At David's suggestion we will be attending superop but would like to get
the basics working in our lab.
Regards,
Steven Gage
Teliris
55 Broadway | NY NY 10006
O+1.212.490.1065 x1400
F+1.212.202.5432
M+1.917.952.2212
------------------------------------------------------------------------
----
This message is a PRIVATE communication. This message and all
attachments are a private communication are confidential or
protected by privilege. If you are not the intended recipient, you are
hereby notified that any disclosure, copying, distribution or use of the
information contained in or attached to this message is strictly
prohibited. Please notify the sender of the delivery error by replying
to this message, and then delete it from your system. Thank you.
________________________________
From: "David Benham (dbenham)" <db...@ci...>
<mailto:db...@ci...>
Date: Wed, 13 Apr 2011 12:58:02 -0700
To: Steven Gage<ste...@te...>
<mailto:ste...@te...>
Subject: RE: Teliris & TIP Protocol License Agreement
Hi Steven
The TIP protocol specification does recommend ("SHOULD"), but does not
mandate, that endpoints default back to a p2p video and audio call if
TIP is not negotiated. The CTS 1000 you have will default back to an
audio only (G.7xx) call, in such a case.
Also, when you do attempt to negotiate TIP with that CTS 1000, you need
to indicate AAC-LD in the SDP audio line (per sect 3.1 in the Cisco TIP
Endpoint Implementation Profile). Otherwise, the CTS endpoint will
assume it is not a TIP session and resort to an audio-only call.
Let us know if this helps or can answer more questions.
From: Steven Gage [mailto:ste...@te...]
Sent: Wednesday, April 06, 2011 6:56 PM
To: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement
Importance: High
David,
I posted our issue, here is a synopsis:
We are trying to setup call between a CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42).
The Mirial client is a pure SIP Video Endpoint capable of supporting
H264 AVC and G711.
We have registered it to the CUCM (version 8.5) as a 3rd party Advanced
SIP Endpoint with Digest authentication
We have a CTS 1000 registered to same CUCM
While trying to setup call between the CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42), we are not able to setup a video call. We see in
the SIP Packet Transfer, that CUCM re-writes (there are 2
SIP Invites sent, but that gets fairly in depth SIP wise) the invite as
audio only invite.
Basically we need to figure out what is the configuration (do we have to
add a SIP Trunk etc) on the CUCM to make it recognize/support video
calls with a pure SIP Video Endpoint.
Thanks,
Steve
On Apr 2, 2011, at 10:36 AM, David Benham (dbenham) wrote:
Steven
Considering it is only 6 weeks away, I strongly suggest attending IMTC's
SuperOp event, where Cisco and Polycom (at least) will have gear set up
to test with each other's TIP products. Non-members are invited to
this SuperOp, but do note that if you join IMTC, you can also partake in
the TIP Activity Group's stewardship of TIP as well as other membership
benefits.
Here are links and email addresses for further query.
IMTC page for TIP Developers
http://www.imtc.org/tip/
IMTC page for TIP Activity Group
http://www.imtc.org/activity_groups/tip.asp
Places to post questions tip...@im...
or at the TIP open source project
http://sourceforge.net/projects/tiprotocol/support
IMTC page for SuperOp event (top item on page)
http://www.imtc.org/events/
Cisco page for TIP Developers
http://www.cisco.com/go/tip
|
|
From: David B. (dbenham) <db...@ci...> - 2011-04-20 19:26:39
|
Aravind
Two issues popped up in our quick review ...
1) Not enough bandwidth in the offer for Video; 320K.
Venturing a guess, have you configured the CUCM's default region
bandwidth?
If not, do the following after logging in to the CUCM
System-->Region, click "find", then the default region will show.
Click "default".
Go to "Modify Relationship to other Regions, and select "Default".
Under Max Video Call Bit Rate, enter "32000", and select the "kpbs"
radio button.
Click "Save", then "Reset".
2) The ACK SIP to our SIP 200 OK only indicates G.711, so be sure you
offer/ack AAC-LD first or only
m=audio 53008 RTP/AVP 0^M
b=TIAS:64000^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M
From: Aravind Sethuraman [mailto:ara...@te...]
Sent: Tuesday, April 19, 2011 2:01 PM
To: David Benham (dbenham)
Cc: ste...@te...; tip...@so...
Subject: Re: Teliris & TIP Protocol Start-up Issues
Importance: High
David,
Thanks for getting back..
As per your request attached are the invite.txt and the cts log files
this time, the behaviour was CTS reports Configuration mismatch....
much obliged
Aravind
On 04/19/2011 04:12 PM, David Benham (dbenham) wrote:
Aravind,
Send us a full copy of your initial SIP INVITE containing the SDP.
Also send a copy of the CTS' logs just in case.
From: Aravind Sethuraman [mailto:ara...@te...]
Sent: Tuesday, April 19, 2011 9:31 AM
To: ste...@te...
Cc: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement
Hello David
Firstly i would like to thank you for offering your help.
What we are building is a TIP user agent plugin which acts as a B2BUA
talking TIP on one side and SIP/H264SVC on the other side. Our ultimate
goal is to build this to interop with Cisco CTS family of products,
1000-3000. We are currently testing with a CUCM version 8.5 and CTS 1000
version 1.7 and we have chosen to adopt TIP profile version 6.
The TIPUA does support the following
1. SIP signaling
2. Video
1. supports H264
2. supports b=TIAS
3. supports RTP format 112 and packetization mode
4. supports the profile-id - 4d0028
3. Audio
1. AAC-LD (we are faking it currently but @ signal level we
are doing it to make CTS happy. We will eventually have this up and
running)
2. G711
4. KeyPress
1. RFC2833 ONLY
5. TIP Negotiation -> We are having the TIP negotiation module
which i will define further below...
What we are seeing is as follows:-
1. Our TIPUA sends invite to CTS (both are registered to CUCM)
2. CTS sends its SDP in 183
3. upon receipt of 183, we start the TIP negotiation -> note, We
are assuming the RTCP port available is RTP+1 (in the SDP)
1. We are using the libTIP version 1.3 library.
4. eventually CTS sends a mid-call re-invite and downgrades to
audio only call.
5. We have not been able to capture ANY RTCP/TIP packets from the
CTS leading to eventual failure of TIP and therefore AUDIO only call.
I am at my wits' end. Are we doing anything glaringly wrong? does CTS
1.7 NOT support TIP ?(highly unlikely)
I am also available for any phone call if you feel it is necessary to
have @ 1917-355-0119
much obliged
Aravind Sethuraman
On 04/13/2011 05:42 PM, Steven Gage wrote:
David,
Aravind is our Chief Software Architect and the technical lead on our
Interop efforts.
Aravind,
David has been extremely helpful in providing direction on TIP
Interoperability.
Please document our assumptions specifically re sip and tip end point
connectivity, minimum bandwidth requirements for video, and audio codec
requirements.
At David's suggestion we will be attending superop but would like to get
the basics working in our lab.
Regards,
Steven Gage
Teliris
55 Broadway | NY NY 10006
O+1.212.490.1065 x1400
F+1.212.202.5432
M+1.917.952.2212
------------------------------------------------------------------------
----
This message is a PRIVATE communication. This message and all
attachments are a private communication are confidential or
protected by privilege. If you are not the intended recipient, you are
hereby notified that any disclosure, copying, distribution or use of the
information contained in or attached to this message is strictly
prohibited. Please notify the sender of the delivery error by replying
to this message, and then delete it from your system. Thank you.
________________________________
From: "David Benham (dbenham)" <db...@ci...>
<mailto:db...@ci...>
Date: Wed, 13 Apr 2011 12:58:02 -0700
To: Steven Gage<ste...@te...>
<mailto:ste...@te...>
Subject: RE: Teliris & TIP Protocol License Agreement
Hi Steven
The TIP protocol specification does recommend ("SHOULD"), but does not
mandate, that endpoints default back to a p2p video and audio call if
TIP is not negotiated. The CTS 1000 you have will default back to an
audio only (G.7xx) call, in such a case.
Also, when you do attempt to negotiate TIP with that CTS 1000, you need
to indicate AAC-LD in the SDP audio line (per sect 3.1 in the Cisco TIP
Endpoint Implementation Profile). Otherwise, the CTS endpoint will
assume it is not a TIP session and resort to an audio-only call.
Let us know if this helps or can answer more questions.
From: Steven Gage [mailto:ste...@te...]
Sent: Wednesday, April 06, 2011 6:56 PM
To: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement
Importance: High
David,
I posted our issue, here is a synopsis:
We are trying to setup call between a CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42).
The Mirial client is a pure SIP Video Endpoint capable of supporting
H264 AVC and G711.
We have registered it to the CUCM (version 8.5) as a 3rd party Advanced
SIP Endpoint with Digest authentication
We have a CTS 1000 registered to same CUCM
While trying to setup call between the CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42), we are not able to setup a video call. We see in
the SIP Packet Transfer, that CUCM re-writes (there are 2
SIP Invites sent, but that gets fairly in depth SIP wise) the invite as
audio only invite.
Basically we need to figure out what is the configuration (do we have to
add a SIP Trunk etc) on the CUCM to make it recognize/support video
calls with a pure SIP Video Endpoint.
Thanks,
Steve
On Apr 2, 2011, at 10:36 AM, David Benham (dbenham) wrote:
Steven
Considering it is only 6 weeks away, I strongly suggest attending IMTC's
SuperOp event, where Cisco and Polycom (at least) will have gear set up
to test with each other's TIP products. Non-members are invited to
this SuperOp, but do note that if you join IMTC, you can also partake in
the TIP Activity Group's stewardship of TIP as well as other membership
benefits.
Here are links and email addresses for further query.
IMTC page for TIP Developers
http://www.imtc.org/tip/
IMTC page for TIP Activity Group
http://www.imtc.org/activity_groups/tip.asp
Places to post questions tip...@im...
or at the TIP open source project
http://sourceforge.net/projects/tiprotocol/support
IMTC page for SuperOp event (top item on page)
http://www.imtc.org/events/
Cisco page for TIP Developers
http://www.cisco.com/go/tip
|
|
From: David B. (dbenham) <db...@ci...> - 2011-04-20 19:13:35
|
Aravind
Two issues popped up in our quick review ...
1) Not enough bandwidth in the offer for Video; 320K.
Venturing a guess, have you configured the CUCM's default region
bandwidth?
If not, do the following after logging in to the CUCM
System-->Region, click "find", then the default region will show.
Click "default".
Go to "Modify Relationship to other Regions, and select "Default".
Under Max Video Call Bit Rate, enter "32000", and select the "kpbs"
radio button.
Click "Save", then "Reset".
2) The ACK SIP to our SIP 200 OK only indicates G.711, so be sure you
offer/ack AAC-LD first or only
m=audio 53008 RTP/AVP 0^M
b=TIAS:64000^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M
From: Aravind Sethuraman [mailto:ara...@te...]
Sent: Tuesday, April 19, 2011 2:01 PM
To: David Benham (dbenham)
Cc: ste...@te...; tip...@so...
Subject: Re: Teliris & TIP Protocol Start-up Issues
Importance: High
David,
Thanks for getting back..
As per your request attached are the invite.txt and the cts log files
this time, the behaviour was CTS reports Configuration mismatch....
much obliged
Aravind
On 04/19/2011 04:12 PM, David Benham (dbenham) wrote:
Aravind,
Send us a full copy of your initial SIP INVITE containing the SDP.
Also send a copy of the CTS' logs just in case.
From: Aravind Sethuraman [mailto:ara...@te...]
Sent: Tuesday, April 19, 2011 9:31 AM
To: ste...@te...
Cc: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement
Hello David
Firstly i would like to thank you for offering your help.
What we are building is a TIP user agent plugin which acts as a B2BUA
talking TIP on one side and SIP/H264SVC on the other side. Our ultimate
goal is to build this to interop with Cisco CTS family of products,
1000-3000. We are currently testing with a CUCM version 8.5 and CTS 1000
version 1.7 and we have chosen to adopt TIP profile version 6.
The TIPUA does support the following
1. SIP signaling
2. Video
1. supports H264
2. supports b=TIAS
3. supports RTP format 112 and packetization mode
4. supports the profile-id - 4d0028
3. Audio
1. AAC-LD (we are faking it currently but @ signal level we
are doing it to make CTS happy. We will eventually have this up and
running)
2. G711
4. KeyPress
1. RFC2833 ONLY
5. TIP Negotiation -> We are having the TIP negotiation module
which i will define further below...
What we are seeing is as follows:-
1. Our TIPUA sends invite to CTS (both are registered to CUCM)
2. CTS sends its SDP in 183
3. upon receipt of 183, we start the TIP negotiation -> note, We
are assuming the RTCP port available is RTP+1 (in the SDP)
1. We are using the libTIP version 1.3 library.
4. eventually CTS sends a mid-call re-invite and downgrades to
audio only call.
5. We have not been able to capture ANY RTCP/TIP packets from the
CTS leading to eventual failure of TIP and therefore AUDIO only call.
I am at my wits' end. Are we doing anything glaringly wrong? does CTS
1.7 NOT support TIP ?(highly unlikely)
I am also available for any phone call if you feel it is necessary to
have @ 1917-355-0119
much obliged
Aravind Sethuraman
On 04/13/2011 05:42 PM, Steven Gage wrote:
David,
Aravind is our Chief Software Architect and the technical lead on our
Interop efforts.
Aravind,
David has been extremely helpful in providing direction on TIP
Interoperability.
Please document our assumptions specifically re sip and tip end point
connectivity, minimum bandwidth requirements for video, and audio codec
requirements.
At David's suggestion we will be attending superop but would like to get
the basics working in our lab.
Regards,
Steven Gage
Teliris
55 Broadway | NY NY 10006
O+1.212.490.1065 x1400
F+1.212.202.5432
M+1.917.952.2212
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From: "David Benham (dbenham)" <db...@ci...>
<mailto:db...@ci...>
Date: Wed, 13 Apr 2011 12:58:02 -0700
To: Steven Gage<ste...@te...>
<mailto:ste...@te...>
Subject: RE: Teliris & TIP Protocol License Agreement
Hi Steven
The TIP protocol specification does recommend ("SHOULD"), but does not
mandate, that endpoints default back to a p2p video and audio call if
TIP is not negotiated. The CTS 1000 you have will default back to an
audio only (G.7xx) call, in such a case.
Also, when you do attempt to negotiate TIP with that CTS 1000, you need
to indicate AAC-LD in the SDP audio line (per sect 3.1 in the Cisco TIP
Endpoint Implementation Profile). Otherwise, the CTS endpoint will
assume it is not a TIP session and resort to an audio-only call.
Let us know if this helps or can answer more questions.
From: Steven Gage [mailto:ste...@te...]
Sent: Wednesday, April 06, 2011 6:56 PM
To: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement
Importance: High
David,
I posted our issue, here is a synopsis:
We are trying to setup call between a CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42).
The Mirial client is a pure SIP Video Endpoint capable of supporting
H264 AVC and G711.
We have registered it to the CUCM (version 8.5) as a 3rd party Advanced
SIP Endpoint with Digest authentication
We have a CTS 1000 registered to same CUCM
While trying to setup call between the CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42), we are not able to setup a video call. We see in
the SIP Packet Transfer, that CUCM re-writes (there are 2
SIP Invites sent, but that gets fairly in depth SIP wise) the invite as
audio only invite.
Basically we need to figure out what is the configuration (do we have to
add a SIP Trunk etc) on the CUCM to make it recognize/support video
calls with a pure SIP Video Endpoint.
Thanks,
Steve
On Apr 2, 2011, at 10:36 AM, David Benham (dbenham) wrote:
Steven
Considering it is only 6 weeks away, I strongly suggest attending IMTC's
SuperOp event, where Cisco and Polycom (at least) will have gear set up
to test with each other's TIP products. Non-members are invited to
this SuperOp, but do note that if you join IMTC, you can also partake in
the TIP Activity Group's stewardship of TIP as well as other membership
benefits.
Here are links and email addresses for further query.
IMTC page for TIP Developers
http://www.imtc.org/tip/
IMTC page for TIP Activity Group
http://www.imtc.org/activity_groups/tip.asp
Places to post questions tip...@im...
or at the TIP open source project
http://sourceforge.net/projects/tiprotocol/support
IMTC page for SuperOp event (top item on page)
http://www.imtc.org/events/
Cisco page for TIP Developers
http://www.cisco.com/go/tip
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