# Unable to do SIP calling between WebRTC clients

**URL:** <https://community.asterisk.org/t/unable-to-do-sip-calling-between-webrtc-clients/70703>\
**Category:** Asterisk Support\
**Created:** [May 11, 2017, 5:32am UTC](https://community.asterisk.org/t/unable-to-do-sip-calling-between-webrtc-clients/70703 "2017-05-11T05:32:37Z")\
**Posts on this page:** 1\
**Showing post:** 83

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**Author:** ![navaismo](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/navaismo/32/2481_2.png) [@navaismo](https://community.asterisk.org/u/navaismo)\
**Post date:** [June 2, 2017, 7:44am UTC](https://community.asterisk.org/t/unable-to-do-sip-calling-between-webrtc-clients/70703/83 "2017-06-02T07:44:08Z")

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When you are using the media gateway you configure the peer as a normal peer. Try reading a little about everything you want to use:

> **[webrtc2sip - Smart SIP and Media Gateway to connect WebRTC endpoints](https://www.doubango.org/webrtc2sip/)**
>
> Smart SIP and Media Gateway to connect WebRTC endpoints

> **[technical-guide-1.0.pdf](https://www.doubango.org/webrtc2sip/technical-guide-1.0.pdf)**
>
> 514.47 KB

> [@Troubleshooting WebRTC Issues](https://community.asterisk.org/t/troubleshooting-webrtc-issues/46011):
>
> Here is a little guide to troubleshoot webrtc issues with Asterisk. Always try to use the latest WebRTC API with the latest Asterisk branch(11 or 12). Be sure you have the icessuport enabled in the rtp.conf ISSUE: I get this response on JSSIP or SIPML5 debug:tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. This issue is caused because you asterisk don’t have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your…

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_[View the full topic](https://community.asterisk.org/t/unable-to-do-sip-calling-between-webrtc-clients/70703)._