Asterisk 11 and sipml5


I make call from SPA502G -> sip,udp -> Asterisk 11 -> sip,ws -> sipml5 , after connected me heard , but I not hear. But sometime work.

If make call from sipml5-> sip,ws -> Asterisk 11 -> sip,udp -> SPA502G then all done always.

What do ?


Try asking on a support forum (this is a discussion one), and providing relevant logging. See … nformation for how to obtain logs.

Are you using the media gateway webrtc2sip or connecting directly to asterisk? Also share the complete sip debug and the JavaScript log from your browser.

Connecting directly to asterisk. Version of Asterisk 11.4.0 without any patch.

Ok, share the requested logs.

Made two try of calls, first good (me heard and I hear) and second bad (me heard , but I not hear)

There is no javascript log(sipml5 log). The network isn’t blocking the ICE servers?

Asterisk and WEB-Server/WEB-Clients working on grey address without NAT


Your JS log has this:

Usually it show as(in my case):

Then you have an error:

==session event = m_stream_audio_remote_added Uncaught TypeError: Cannot set property 'src' of null call.js:564 To be honest i don’t know if that is part of your issue but comparing with my workings js log I don’t have those.

Are you using your own code or using the sipml5 example page?

this error is part of my code.

because working without nat I remove ice servers.

any other problems ?

May be trouble with Chrom version , which work without problems ?

As far I know ICE servers are needed by SIPML5, latest chrome version works and you need to debug deeper, with js and asterisk side like network properties, Nat & RTP debug. Finally test the sipml5 original demo, if it works then you need to take a look at your code.

  1. On the demo site problems were also ( I suppose the problem with NAT, therefore made test in local area )
  2. I testing without any NAT , all in local area ( In this case ice servers need ? )
  3. On asterisk side i don’t see any problems , on JS side look good

Take a look on this thread on it explain how it works, also sear h in the doubango discussion group for similar issues!msg … EtCNcu1mMJ

In the group many people has issues when they edit the original code & have bad Nat settings on asterisk side.

In asterisk rtp debug i see that traffic going only to one side (to web-phone), from web-softphone RTP does not go to Asterisk.

Also sometime all work without any change.

May be at that moment asterisk 11 (without any patch) and sipml5 don’t work normally.

Therefore I installed webrtc2sip and all working as expected.