[Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

[Tarak Patel]

Hi Andrew,

I was able to make a successful call between sip client --> webRTC client, But with only audio media. Somehow, video does not come.

Can you guide me on how to have full video call between sip --> webRTC?

Calls from webRTC Client --> Sip client fails. 

Let me know, if I can able to do above scenario with sipwise.

Note: My sip client does not have support for vp8. I can only use h.264, h.263plus and h.263 video codecs.

Thanks,
Tarak Patel

-----Original Message-----
From: Spce-user [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Tarak Patel
Sent: Friday, April 10, 2015 1:03 PM
To: Andrew Pogrebennyk; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

Hi Andrew,

MR3.8.1 Fixed my issues with OnSip Sip.Js now it works like charm. 

Also May I know how can I configure sipwise so that I can route call from webrtc to legacy sip?

Thanks, for all the efforts and helping me out solving this problem. I really appreciate it.

Thanks,
Tarak Patel

-----Original Message-----
From: Spce-user [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Tarak Patel
Sent: Tuesday, March 24, 2015 1:04 PM
To: Andrew Pogrebennyk; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

Hi Andrew, 

Please find answers to your questions below.

1) Could you send me the Chrome version and OS version you are using?

--    I am running windows 7 and chrome version 41.0.2272.101m
--    For sipwise I have just update the vm for mr3.7.2 to get your latest changes.

2) is there any way I can test the onsip library with my spce machine?
-- Definitely you can test it locally. Their on your end to. I can send you the test application.
-- All you have to do is install node on your computer and run app using node. Right now I am using node to deploy the app.
-- After installing node, all you have to do is migrate to the root directory of the above app. You will find app.js. Their run the command node app.js
-- After this you can access the web app using <local-ip-address>:8083/caller.html for caller and receiver.html for receiver.
-- Please Download the app from below link,
https://www.dropbox.com/s/9a1ent129rfq3t9/webAppTestGround.rar?dl=0

3) could you send me your
/etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.customtt.tt2
?
--Please find the attachement here. 

Let me know this helps you out or not.

Thanks,
Tarak Patel

-----Original Message-----
From: Andrew Pogrebennyk [mailto:apogrebennyk at sipwise.com]
Sent: Tuesday, March 24, 2015 12:42 PM
To: Tarak Patel; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

Hi Tarak,

I made successful tests using sipml5 client in Chromium with the provided instructions (using "red" as video codec, not vp8). The firefox had the issues like you described (no remote video).

We're still taking the plunge and change the user preferences and add selection between rtp/savpf, udp/tls/rtp/savpf etc in the upcoming
mr3.8.1 (which will be released very soon), but this is not necessarily connected to the issues you were having.

Yeah, I'd like to have some further information..
1) Could you send me the Chrome version and OS version you are using?
2) is there any way I can test the onsip library with my spce machine?
3) could you send me your
/etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.customtt.tt2
?

Cheers,
Andrew

On 03/24/2015 05:31 PM, Tarak Patel wrote:
> Hi Andrew,
> 
> Let me know if there is any progress on the underlying issues with Onsip sip.js Webrtc and sipwise.
> 
> If its fixed, How can I obtain the latest copy? Also, Let me know if you required further information or try something different.
> 
> I will be happy to provide any such information to  you and test different scenarios if you want.
> 
> Thanks,
> Tarak Patel
> 
> -----Original Message-----
> From: Spce-user [mailto:spce-user-bounces at lists.sipwise.com] On Behalf 
> Of Tarak Patel
> Sent: Thursday, March 19, 2015 10:15 AM
> To: Andrew Pogrebennyk; spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise
> 
> Hi Andrew,
> 
> Both those parameters are set to transparent in the domain. Let me know, if there is any progress related to the issue.
> 
> Once again thanks spending your valuable time in helping out me.
> 
> Thanks,
> Tarak Patel
> 
> -----Original Message-----
> From: Andrew Pogrebennyk [mailto:apogrebennyk at sipwise.com]
> Sent: Thursday, March 19, 2015 7:39 AM
> To: Tarak Patel; spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise
> 
> Hi,
> are you sure that the preferences srtp_transcoding and rtcp_feedback are set to Transparent?
> 
> Maybe the patch just doesn't work as I described, I have to put more testing in webrtc in the coming days. I will let you know!
> 
> Regards,
> Andrew
> 
> On 03/18/2015 06:33 PM, Tarak Patel wrote:
>> Sorry about those silly mistakes by me. 
>>
>> But still I am getting same results. With Local Preview only on both the ends.
>>
>> I have attached the logs as well now.
>>
>> Thanks,
>> Tarak Patel
> 
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> Spce-user at lists.sipwise.com
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> 

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