[SIP Beyond VoIP] Missing incoming RTP stream

[Adrian Georgescu]

You should know  that the STUN server setting is used in the context of ICE negotiation.  Unless you use ICE, that setting is useless, it does not do anything.

Adrian


On Jun 26, 2013, at 1:21 PM, Janusz Kowalczyk <kowalczykjanusz at gmail.com> wrote:

> Thanks Saul. Will try to use this script as a base for my script :)
> 
> I'm not sure whether I have to use STUN server. 
> I tried using "sip-session" or "sip-audio-session" script with and without STUN server configured and RTP worked both ways. And those calls were made to/from EC2 instance.
> 
> This is slightly out of topic, but the only problem I have with both of those scripts is that they send DTMF as RTP events even if the account is configured to use inband ones. 
> I've enabled inband dtmf by:
> sip-settings -a set user at myaccount.com rtp.inband_dtmf=True
> 
> To configure a STUN server for my account [ I used one out of many listed here http://www.tek-tips.com/faqs.cfm?fid=7542 ] I run:
> sip-settings -a set user at myaccount.com nat_traversal.stun_server_list='stun.ekiga.net'
> 
> Here's how I launch the scripts:
> sip-session --auto-answer -S -c ~/.sipclient -a user at myaccount.com
> sip-audio-session --auto-answer -S -c ~/.sipclient -a user at myaccount.com
> 
> When I use "sip-session" to call my desk phone, and I use "/dtmf {0-9}" command, then on desk phone I can barely hear some really faint and really short signal which presumably is the DTMF.
> When I use "sip-audio-session" to answer incoming calls from my desk phone, when I press keys on my phone then all tones are send as RTP events.
> 
> I could attach pcap files but they're stripped off automatically :)
> 
> Here's my config file: ~/.sipclient/config
> Accounts:
>     user at myaccount.com
>         enabled = true
>         auth:
>             password = some_password
> 
>         nat_traversal:
>             stun_server_list = "stun.ekiga.net:3478",
> 
>         rtp:
>             audio_codec_list = PCMA,
>             inband_dtmf = true
> 
>         sip:
>             outbound_proxy = "nat.myaccount.com:5065;transport=udp"
> 
> SIPSimpleSettings:
>     default_account = user at myaccount.com
>     instance_id = "urn:uuid:e83fa8c5-80b1-482f-bb2a-e9dd7bdd9ef8"
>     audio:
>         alert_device =
>         input_device =
>         output_device =
> 
> 
> 
> 
> 
> 
> 
> 
> On 25 June 2013 17:31, Saúl Ibarra Corretgé <saul at ag-projects.com> wrote:
> 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
> AG Projects
> 
> 
> 
> 
> 
> 
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