[SIP Beyond VoIP] Missing incoming RTP stream

[Saúl Ibarra Corretgé]

Hi,

On Jun 25, 2013, at 4:55 PM, Janusz Kowalczyk wrote:

> Hi Guys,
> 
> I'm trying to use SipSimple SDK write test scripts that will be executed with fabric on a remote machine without a soundcard (ie. Amazon EC2) and which would dial in to our voice service and play back some wav files and/or DTMF's. 
> 
> 
> At the moment I'm trying to use modified version of Saul's "jamesbond" script
> https://github.com/saghul/sipsimple-examples/tree/master/jamesbond
> to make a call and play a wav file once call is established. Modified script is at the end of email.
> 
> The problem is that there's no incoming RTP stream. I doubled check firewall rules and UDP traffic is allowed on all ports. I've attached a pcap file from an example call.
> 
> I'd be grateful if you could help me to figure out what's the problem.
> 

Unless your service does NAT traversal you'll not be able to get inbound RTP, because you are running from a private IP address, unless there is direct routing between the two endpoints, which I'm assuming there isn't.

> btw. Is there a good and simple example script that would make unattended calls, play wav files and DTMF tones?
> 

Have a look at sip-audio-session script, it's part of the sipclients package. It doesn't play DTMF, but shouldn't be hard to add.


Cheers,

--
Saúl Ibarra Corretgé
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