what is default rtp port range

[Metehan Metin]

I was checking janus.plugin.sip.cfg file and i saw this lines at below

;Range of ports to use for RTP/RTCP (default=10000-60000)
rtp_port_range = 20000-40000

I just set up janus v0.2.5 and i didn't change anything.
i confused which one is default 10000-60000 or 20000-40000

[Mirko Brankovic]

I guess the janus.cfg is important one, so [media] section

[media]
	;rtp_port_range = 20000-40000

but if you don't get anything set, I guess, jugging by the comment, a full range of 0 to 65535 will be used 

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Regards,

Mirko

[Metehan Metin]

8 Aralık 2017 Cuma 15:30:20 UTC+3 tarihinde Mirko Brankovic yazdı:

I guess the janus.cfg is important one, so [media] section

[media]
	;rtp_port_range = 20000-40000

but if you don't get anything set, I guess, jugging by the comment, a full range of 0 to 65535 will be used 

On Fri, Dec 8, 2017 at 12:42 PM, Metehan Metin <mete...@gmail.com> wrote:

I was checking janus.plugin.sip.cfg file and i saw this lines at below

;Range of ports to use for RTP/RTCP (default=10000-60000)
rtp_port_range = 20000-40000

I just set up janus v0.2.5 and i didn't change anything.
i confused which one is default 10000-60000 or 20000-40000

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Regards,

Mirko

In fact i am living one way audio problem some calls(not every call). My sip santral rtp port range is between 10000-12000.
 i think problem causing because of this. I want to be sure janus and my sip santral using same rtp port range for media.
Do u know how can i check which rtp port is using for media in a call?(Should i look janus gateway logs?)

[Mirko Brankovic]

From browser point of view, easiest is to use chrome://webrtc-internals/

and look for the:

Conn-audio-1-0 (googCandidatePair)

There will be more than 1-0, for each candidate one, but bolded should be the active one that is sending and receiving packets

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Regards,

Mirko

[Metehan Metin]

8 Aralık 2017 Cuma 16:40:10 UTC+3 tarihinde Mirko Brankovic yazdı:

I checked Conn-audio-1-0 (googCandidatePair) and i saw <janus_gw_ip>:40593(etc.)
Then i changed janus.plugin.sip.cfg rtp_port_range = 1000-12000 and didn't change anything.

But when i tried Janus.cfg [media] rtp_port_range = 1000-12000, it worked.
The port was <janus_gw_ip>:10000. I hope it fixes my one way audio problem.

Thanks for Help  Mirko

[Mirko Brankovic]

Np hope it helped ;)

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[Lorenzo Miniero]

I think you guys are talking of different things, since the SIP plugin was mentioned. When the SIP plugin is used, there are two different RTP ranges: the one for WebRTC users, and the one for RTP on the SIP side. They are NOT the same thing. You configure the former in janus.cfg, and the latter in janus.plugin.sip.cfg. The default for both, when nothing is specified, is "use whatever port is available", or whatever the OS is configured to return when asking for a random port.

L.

[Metehan Metin]

11 Aralık 2017 Pazartesi 13:30:18 UTC+3 tarihinde Lorenzo Miniero yazdı:

I tought SIP and Webrtc ports configuring  janus.cfg . In fact i want to configure sip rtp ports. Because i am living one way audio problem.
Should i change  janus.plugin.sip.cfg for sip ports ?
Also could i check with admin api which sip ports using for media  ?

[Lorenzo Miniero]

Il giorno lunedì 11 dicembre 2017 13:46:00 UTC+1, Metehan Metin ha scritto:

I tought SIP and Webrtc ports configuring  janus.cfg . In fact i want to configure sip rtp ports. Because i am living one way audio problem.
Should i change  janus.plugin.sip.cfg for sip ports ?

Yes.

 

Also could i check with admin api which sip ports using for media  ?

No, you won't find info on those ports in the Admin API, as it's a plugin specific thing and the SIP plugin doesn't add them. If you want that, you'll have to modify the query_session callback in the SIP plugin code so that the ports are listed too.

L. 

[Mirko Brankovic]

Just do the tcpdump or ngrep and see what was negotiated

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[Metehan Metin]

11 Aralık 2017 Pazartesi 16:33:31 UTC+3 tarihinde Lorenzo Miniero yazdı:

I editted janus.sip.c. I added a log line.

https://github.com/metehanm4/janus-gateway/blob/master/plugins/janus_sip.c#L3520-L3522

I logged local_audio_rtp_port as error to see easly.

When i change janus.plugin.sip.cfg rtp port the local_audio_rtp_port was changeing.

Thanks for help Lorenzo