[Spce-user] Fw: webRTC in production

[Andreas Granig]

Hi,

>From a general point of view, you can just point your WebRTC clients to
https://<ip>/wss/sip/, which is internally redirected to kamailio-lb at
https://<ip>:5061/ws. That's it from a signalling point of view.

In regards to media, the SPCE can be configured for example in your
peering server and domain preferences to force plain RTP for your SIP
interconnects for PSTN origination and termination, most notably the
srtp_transcoding and rtcp_feedback preferences. That way, the rtpengine
can pick up DTLS-SRTP on your local subscriber side and transform it to
plain RTP on the peering side and vice versa.

Andreas

On 11/17/2014 10:36 AM, H Yavari wrote:
> 
> Hi,
> Thanks for your reply. Are there any Docs and etc. for implementation?
>  
>  
> Regards,
> H. Yavari
> 
>> ------------------------------------------------------------------------
>>
>> Hi,
>>
>> Within the Sipwise platforms, we're using kamailio for
>> SIP-over-Websocket to do the signalling and rtpengine for the SRTP-DTLS
>> transcoding, which is considered a highly performing and robust combo
>> being in use by really big Telcos in production today.
>>
>> Andreas
>>
>> On 11/17/2014 09:33 AM, H Yavari wrote:
>>> Hi,
>>>
>>> I read some posts like TV over webRTC in the Sipwise website. I'm so
>>> interested to implement webRTC with this. want to know that this is an
>>> robust solution and it can be used in production?  because the Asterisk
>>> and FS have problem with SRTP-DTLS and not support webRTC fully yet.
>>>
>>> So any one have experience with webRTC implementation with Sipwise in
>>> production?
>>>
>>>
>>> Regards,
>>> H. Yavari
>>>
>>>
>>>
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