[Blink] Calling Blink To Blink client Via Opensips server , results to "media stream timed out while starting"

[Dan Pascu]

On 1 Apr 2015, at 23:36, the.ozma.of.oz the.ozma.of.oz wrote:

> Hello. First of all i want to say that i'm really enjoying the blink softphone, the features that you made are awesome! But, i'm stuck on the weird issue for 2 months now.
> 
> I have 2 accounts: bob and alex. When i do call bob from alex i get this kind of behaviour in the bob's softphone:
> 
> 
>     http://s017.radikal.ru/i404/1504/37/f067401e43b0.png
> 
> 
>   It hangs on "starting media" for 15 seconds all that time. On the other end, in alex softphone the connection displayed as established. After that:
> 
>         http://i016.radikal.ru/1504/93/4f262c592fb1.png
> 
> 
> "media stream timed out while starting" in the bob's softphone.
> 
> On the alex softphone side:
> 
> 
>      http://s017.radikal.ru/i413/1504/96/70013a84ccfa.png
> 

There are a couple of things that can result in this:

1. You have a router that implements SIP ALG. I have yet to find a SIP ALG implementation that works. All of them just break your SIP calls. Check your router and disable it if it has it enabled. If you cannot disable it, you can avoid it by using TLS to connect to the servers.

2. It's possible that you have ICE enabled in Blink and your SIP proxy doesn't deal with it properly. You can try disabling ICE (it's a per account setting) and see if it helps, but you need your provider to have a far NAT traversal solution else calls will fail.

> I posted this issue on opensips mailing list, and no one could of helped me. I changed a couple of servers and hostings in different countries, the issue is still there!

Try to create sip2sip accounts (you can create one directly from blink) and use them to test. Our SIP proxies support ICE properly and if you still have the problem with them it might be that you have some SIP ALG router that messes up your traffic. Also on sip2sip you can point us to the call trace and we can see all the SIP traffic from your call and identify the problem.

> But a couple days we tried to call from "Blink" to "PhonerLight" and then from "PhonerLight" to "Blink" client using our opensips server. And the call was successful, media stream didn't time out, we can hear each other.

The 15 seconds until the call timed out and was closed might suggest that the ACK was not received (the call times out if ACK is not received in 15 seconds and the audio on the destination side will not even start until ACK is received). This would support hypothesis #1 above as a SIP ALG router might break some of your traffic in a way that it makes it not arrive at the destination.

But the fact that you can have a call between blink and another client might suggest that the problem might be #2 above as most likely the other client doesn't implement ICE.

> So i decided to write to this mailing list, maybe you can help me to solve this issue. If needed i can provide the blink logs.

The logs could help clarify which of the 2 it is (or if maybe something else entirely). If you send them and want to keep them private send them to support at ag-projects.com, but please do not obfuscate any IP addresses in them as we won't be able to tell what the problem is if they are all xx.xx.xx.xx and yy.yy.yy.yy. Also we would need the logs from both ends to identify what is going on. I also suggest to open preferences and use "Clear log files" right before you make the test call so it will only contain the SIP traffic from the test call.

I would still suggest you try to use sip2sip.info accounts to test things as you can see all your calls in a web interface and point us to the failed call for which we will have the server logs that are more useful than the client logs as we can see if packets were mangled by SIP ALG routers.

--
Dan