Thanks for the information.

What version of asterisk?

A siptrace will still be helpful.

On 12/11/10, William Russell <[email protected]> wrote:
> I agree that sipXbridge can be codec agnostic for the reason you said. I
> don't think the codec is a problem. The problem is the media TYPE -the
> bridge only parses the SDP for AUDIO media attributes, not VIDEO or IMAGE
> media attributes. Thus it'll never know what port to open up for the video.
> And the audio port is in the same range but NOT the same port. The gateway
> is an asterisk server that we use to decide where the call should be sent
> (similar to a Class4 switch) - depending on if it is an internal number or
> external number. If it is external it is sent to our Sip Trunk provider, if
> not it is sent to the appropriate sipX PBX. This is working correctly as the
> call goes there - the issue is with the video which tries to send it but the
> bridge is denying it.
>
> And I believe the problem is in the SipUtilities.java file. The utility used
> to gather port information only collects for AUDIO, not VIDEO media
> attributes.
> Sent from my Sigma Lambda BetaBerry
>
> -----Original Message-----
> From: Tony Graziano <[email protected]>
> Date: Sat, 11 Dec 2010 10:11:24
> To: <[email protected]>
> Subject: Re: [sipx-dev] Video support in sipXbridge?
>
> Sipxbridge is able to be codec agnostic by removing all codecs (it simply
> passes through all).
>
> What you have not done is provided what type of gateway you are using
> (itsp)
> or if it is sip only call.
>
> Wireshark traces might be helpful, but they do provide internal signalling
> information "that_will_not_show_in_a_pcap_file".
>
> That being said, indicate "how you are dialing" (or being dialed), and a
> siptrace would be helpful.
>
> Since you don't have access over the weekend to do a trace, just indicate
> how you are dialing. Who/what is the gateway?
>
> To be fair, the whole skype thing is an oddity to skype, not to h.264 at
> all.
> ============================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
>
> Email: [email protected]
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> ----- Original Message -----
> From: William Russell <[email protected]>
> To: Tony Graziano <[email protected]>
> Sent: Sat Dec 11 10:03:41 2010
> Subject: Re: [sipx-dev] Video support in sipXbridge?
>
> Unfortunately, I don't have access to this over the weekend, but I can tell
> you that I've looked at the traces (through Wireshark) and the point of
> denial is the bridge. There is NO H.264 RTP media packets leaving the
> bridge
> on either endpoint (both identical sipX configurations and phone
> configurations). The G711 RTP packets pass through with ease, but the H.264
> RTP packets sent from the phone are denied by the bridge with a ICMP port
> unreachable error. I think I've found why there is an error. There is even
> a
> bug sent in around June 6th 2010 that states that because of Skype for Sip
> offering video in 2011 - the bridge must begin to support video. I know
> where the problem is and as soon as I get to my computer I could tell you
> exactly what java file causes this occurrence. I do not think this a major
> change - actually, it's really simple because EVERYTHING else works
> perfectly (signalling, voice traffic, even bye messages).
> Sent from my Sigma Lambda BetaBerry
>
> -----Original Message-----
> From: Tony Graziano <[email protected]>
> Date: Sat, 11 Dec 2010 02:36:00
> To: <[email protected]>; sipXecs developer
> discussions<[email protected]>; <[email protected]>
> Subject: Re: [sipx-dev] Video support in sipXbridge?
>
> A siptrace would be helpful.
>
> Can you peovide the call flow information as it relates to the
> gateway? i.e. Is it going through a pstn or itsp provider?
>
> The siptrace would help define what codecs are being offered/accepted.
> A call using h.264 uses the same media ports as a voice only call. So
> it could be a gateway or endpoint that is not offering or allowing
> video. Hence the siptrace.
>
> On 12/10/10, William Russell <[email protected]> wrote:
>> Endpoints: Polycom VVX 1500s
>> Protocol: H264
>> There is only one issue: the port that the media is sent to is NOT being
>> opened because the bridge is NOT looking for "video" media attributes
>> such
>> as the port it is being sent to from the phone.
>>
>> I looked into the sipXbridge code. There is a point where the bridge is
>> parsing the SDP for information on the RTP streams - only problem is it
>> is
>> ONLY looking for audio. Also, another java file states ONLY voice codecs
>> and
>> NOT any video codecs.
>>
>> I just wanted to know if I'm on the right path to solving this problem.
>> We
>> are developing a unique call manager system and video must be allowed
>> over
>> the bridge...I think that solving those two problems will fix it.
>>
>> If I'm still not making much sense, could someone explain (specifically)
>> how
>> the bridge (or any other parts of sipxecs) knows WHAT ports to open up
>> for
>> the audio and video streams? That's the real crux of the matter here - if
>> I
>> understand all the moving parts, I can figure out how to make it work for
>> us.
>>
>> Thanks for all your help!
>> Sent from my Sigma Lambda BetaBerry
>>
>> -----Original Message-----
>> From: "Todd Hodgen" <[email protected]>
>> Sender: [email protected]
>> Date: Fri, 10 Dec 2010 18:09:33
>> To: 'sipXecs developer discussions'<[email protected]>
>> Reply-To: [email protected],
>>      sipXecs developer discussions <[email protected]>
>> Subject: Re: [sipx-dev] Video support in sipXbridge?
>>
>> SipXecs certainly supports video, as it is just a media stream.   You can
>> do
>> a video call from a Nortel Phone to a Nortel Phone, or to a Counterpath
>> phone.  You can do a counterpath to Polycom as well, using video.
>>
>> What I haven't seen explained here is what video you are trying to send.
>> What are your endpoints exactly, and what protocol are they expecting,
>> and
>> what are the network requirements?  If they are sip based products, then
>> the
>> question is what ports do they require from end to end?
>>
>> You are hard pressed to find many conference systems that use the PSTN
>> for
>> video, unless it is using ISDN PRI or BRI.  And, those ports are bonded
>> typically to a higher rate than 64k.
>>
>> I think a bit more detail of what you are trying to do will shed some
>> light
>> on what the technical issues are that need to be overcome, if they can
>> be.
>>
>> In the mean time, you can use some of the proven products - Polycom
>> VVX1500,
>> Nortel, and Counterpath.  I did see someone playing with Tanberg on here
>> a
>> few weeks ago as well.
>>
>> -----Original Message-----
>> From: [email protected]
>> [mailto:[email protected]] On Behalf Of Tony Graziano
>> Sent: Friday, December 10, 2010 5:27 PM
>> To: sipXecs developer discussions
>> Subject: Re: [sipx-dev] Video support in sipXbridge?
>>
>> I normally never have a problem with video.
>>
>> I never find I can call with video through a itsp. I normally have to
>> make a sip-to-sip call or setup a site-ti-site dialing plan.
>>
>> Have you tried to do a sip-to-sip call or site dialing planv
>>
>> On 12/10/10, W. E. W. Russell <[email protected]> wrote:
>>> Yeah, I did that, but still no go.
>>> I was looking into the code and in the RTP*.java files under the bridge
>> and
>>> I get the gist that it doesn't really pay any attention to video codecs.
>>> And what I mean by not pay any attention and why that would be an issue
>>> is
>>> that we are seeing that "port unreachable" ICMP error. Since it isn't
>> paying
>>> attention, it never figures out the port it needs to open - I think,
>> correct
>>> me if I'm wrong?
>>>
>>> If copy the section that does it for the audio and make it do the same
>>> for
>>> video as well as allow the video codec, that should fix the problem?
>>>
>>> FYI - All the signalling is good and we can hear the voice just fine, no
>>> problem.
>>>
>>> Best regards,
>>> William
>>>
>>> On Fri, Dec 10, 2010 at 7:22 PM, Tony Graziano
>> <[email protected]
>>>> wrote:
>>>
>>>> Have you tried to manually allow the protocol in the allowed list in
>>>> siptrunking (server>services>sip trunking? Or just to remove all
>>>> codecs to allow all?
>>>>
>>>>
>>>>
>>>> On 12/10/10, W. E. W. Russell <[email protected]> wrote:
>>>> > All,
>>>> >
>>>> > We are trying to make a video call to another PBX and it seems that
>>>> > the
>>>> > voice is fine, but there is no video.
>>>> >
>>>> > I did some research and since we were creating a gateway that uses
>>>> > sipXbridge to call it seems that there are some ports that sipXbridge
>> is
>>>> > blocking with ICMP "port unreachable" issues.
>>>> >
>>>> > Again the voice is fine, but the video is getting blocked before it
>> ever
>>>> > leaves the PBX. Thank you for any help you can provide or direction I
>>>> > can
>>>> go
>>>> > in!
>>>> >
>>>> > Best regards,
>>>> > William
>>>> >
>>>> > --
>>>> > W. E. W. Russell
>>>> > Member of Technical Staff (System Integration) at incNETWORKS, Inc.
>>>> > Alumni of the University of Florida - Fall 2007 - BS in Computer
>> Science
>>>> > Active Alumni member of Sigma Lambda Beta International Fraternity,
>> Inc.
>>>> > Cell Phone # 732-744-6483
>>>> > Work Phone # 732-483-1511
>>>> > Blackberry Pin: 30E9FD36
>>>> >
>>>>
>>>> --
>>>> Sent from my mobile device
>>>>
>>>> ======================
>>>> Tony Graziano, Manager
>>>> Telephone: 434.984.8430
>>>> sip: [email protected]
>>>> Fax: 434.326.5325
>>>>
>>>> Email: [email protected]
>>>>
>>>> LAN/Telephony/Security and Control Systems Helpdesk:
>>>> Telephone: 434.984.8426
>>>> sip: [email protected]
>>>>
>>>> Helpdesk Contract Customers:
>>>> http://support.myitdepartment.net
>>>>
>>>> <http://support.myitdepartment.net>Blog:
>>>> http://blog.myitdepartment.net
>>>>
>>>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>>>> _______________________________________________
>>>> sipx-dev mailing list
>>>> [email protected]
>>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>>>
>>>
>>>
>>>
>>> --
>>> W. E. W. Russell
>>> Member of Technical Staff (System Integration) at incNETWORKS, Inc.
>>> Alumni of the University of Florida - Fall 2007 - BS in Computer Science
>>> Active Alumni member of Sigma Lambda Beta International Fraternity, Inc.
>>> Cell Phone # 732-744-6483
>>> Work Phone # 732-483-1511
>>> Blackberry Pin: 30E9FD36
>>>
>>
>> --
>> Sent from my mobile device
>>
>> ======================
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: [email protected]
>> Fax: 434.326.5325
>>
>> Email: [email protected]
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> sip: [email protected]
>>
>> Helpdesk Contract Customers:
>> http://support.myitdepartment.net
>>
>> <http://support.myitdepartment.net>Blog:
>> http://blog.myitdepartment.net
>>
>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>> _______________________________________________
>> sipx-dev mailing list
>> [email protected]
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>
> --
> Sent from my mobile device
>
> ======================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: [email protected]
> Fax: 434.326.5325
>
> Email: [email protected]
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: [email protected]
>
> Helpdesk Contract Customers:
> http://support.myitdepartment.net
>
> <http://support.myitdepartment.net>Blog:
> http://blog.myitdepartment.net
>
> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>

-- 
Sent from my mobile device

======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
Fax: 434.326.5325

Email: [email protected]

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]

Helpdesk Contract Customers:
http://support.myitdepartment.net

<http://support.myitdepartment.net>Blog:
http://blog.myitdepartment.net

Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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