Unable to do SIP calling between WebRTC clients

Hi,

We are unable to do SIP calling between WebRTC clients. Please help us to resolve the issue.

Thanks in advance.

We are getting errors shown in attached image.

Hi,

Now we are getting incoming call to the WebRTC client but outgoing calls are not happening from WebRTC client. We are getting below errors while doing outgoing calls.

chan_sip.c:26422 handle_request_invite: Call from ‘’ (23.239.84.99:5108) to extension ‘700972592620692’ rejected because extension not found in context ‘public’.

chan_sip.c:10815 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio

Please give suggestions to resolve this issue.

This message can only be produced for incoming calls and is self explanatory, and not specifically related to WebRTC, SIP, or any other channel technology.

The other one will require you to provide details of your configuration and SIP debugging containing the SDP exchange. Please provide these as plain text, marked as pre-formatted for the forum, as screenshots are difficult to search and involve more work to access.

Hi David,

Please help me how to overcome those errors. We are getting incoming calls but after few seconds call getting terminated. Below are the errors we are getting for incoming calls.

– Called SIP/6000
– SIP/6000-00000007 is ringing
[May 11 13:31:28] WARNING[31486][C-00000005]: channel.c:5639 set_format: Unable to find a codec translation path: (opus) -> (ulaw)
[May 11 13:31:28] WARNING[31486][C-00000005]: channel.c:5639 set_format: Unable to find a codec translation path: (ulaw) -> (opus)
– SIP/6000-00000007 answered SIP/6002-00000006
[May 11 13:31:28] WARNING[31537][C-00000005]: channel.c:6538 ast_channel_make_compatible_helper: No path to translate from SIP/6000-00000007 to SIP/6002-00000006
[May 11 13:31:28] WARNING[31537][C-00000005]: app_dial.c:3196 dial_exec_full: Had to drop call because I couldn’t make SIP/6002-00000006 compatible with SIP/6000-00000007

Install an Opus codec.

Hi David,

Please tell me how to install Opus codec and where to install that.

I’ve never used it. The messages tell me that you have nothing installed capable of transcoding Opus and I don’t need to the details of the codec module to determine that.

If it is a non-proprietary module, make sure you enable it in menuselect when you build Asterisk, and in modules.conf when you load Asterisk.

If it is proprietary see its documentation.

In either case, it may have configuration parameters that may need adjusting, or your client may be requesting options that the Asterisk module doesn’t support.

You may find this useful; http://blogs.digium.com/2016/09/30/opus-in-asterisk/

On a quick skim, it looks like it need Asterisk 14.1+ and it is only available for Intel machines (and possibly only provided as a binary).

PS Looking at the sorts of problems people have with WebRTC, it is not something that an Asterisk newbie should be tackling.

Hi David,

Nipun*CLI> module show like res_format_attr_opus.so
Module Description Use Count Status Support Level
res_format_attr_opus.so Opus Format Attribute Module 1 Running core
1 modules loaded

I think opus module was loaded.

That module does not provide transcoding. It is strictly for understanding the parameters in SDP. The blog posted linked is regarding the transcoding module.

Hi Jcolp,

Please help me to resolve the issue.

We are getting incoming calls but call was not establishing. we are getting below errors in asterisk cli.

– Called SIP/6000
– SIP/6000-00000007 is ringing
[May 11 13:31:28] WARNING[31486][C-00000005]: channel.c:5639 set_format: Unable to find a codec translation path: (opus) -> (ulaw)
[May 11 13:31:28] WARNING[31486][C-00000005]: channel.c:5639 set_format: Unable to find a codec translation path: (ulaw) -> (opus)
– SIP/6000-00000007 answered SIP/6002-00000006
[May 11 13:31:28] WARNING[31537][C-00000005]: channel.c:6538 ast_channel_make_compatible_helper: No path to translate from SIP/6000-00000007 to SIP/6002-00000006
[May 11 13:31:28] WARNING[31537][C-00000005]: app_dial.c:3196 dial_exec_full: Had to drop call because I couldn’t make SIP/6002-00000006 compatible with SIP/6000-00000007

@david551 already provided the information to solve that. If you need transcoding from or to opus you have to install the opus transcoding module. You can find it in the “Codec Translators” section of “make menuselect”, or you can choose not to use opus on the browser. And if you are entering the world of WebRTC you really need to learn how to support and debug things yourself, as that happens often when it comes to WebRTC.

@jcolp Please find the below image of menuselect.

@jcolp And I have downloaded the codec_opus but need help how to install that.

You need to ensure you have the required dependencies to automatically download and install it. They are listed below. You are probably missing xmlstarlet.

You need to resolve the missing dependencies, which, for some reason, seem to have been obscured by the buttons.

There is a README[1] directly in the download directories. It provides alternative install instructions. In the future please try to figure this out yourself and look a bit instead of immediately jumping here.

[1] http://downloads.digium.com/pub/telephony/codec_opus/asterisk-13.0/x86-64/README

Where to install xmlstarlet means in which folder.

Wherever binaries are placed on the system. It’s commonly available as a package. In the future if you have specific questions after trying things I’ll look into answering, but right now it feels as though I’m doing this work for you so I’m going to excuse myself for the moment. I’ll let others help if they choose.