[Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise
Andrew Pogrebennyk
apogrebennyk at sipwise.com
Tue Mar 24 12:42:15 EDT 2015
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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