Hi Segriu

 

I think my issue is with  rtpengine . I’m using direction parameter to set a LAN and WAN IP on the offer and I think it’s getting messed up during re-invites

 

 

 

 

 

Best Regards

 

Gerry Kernan

 

From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Sergiu Pojoga
Sent: 23 March 2018 01:34
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: <UNJUNKED> Re: [SR-Users] Audio stops after resuming call from hold

 

OMG, what are the odds, a client reported the same problem today! Edge proxy running same 4.2.3, requests are forwarded to a farm of Asterisks v13 in a similar way based on $rd, far-end NAT traversal is handled by Kamailio.

 

I've had only an hour or so to debug today. Re-invites containing SDP are handled the same way as invites in terms of SDP mangling, all looks good in that sense. There's nothing special to be done about re-invites.

 

Preliminary clue is that this happens (or not) depending on the type of firewall/NAT behind which the phone is located. In the case with the trouble, it's a Sonicwall, probably a Symmetric NAT. Is doesn't happen to a phone behind a Full/Restricted Cone NAT. 

 

What nat= are you setting for Asterisk peers?

Do you engage rtpproxy/rtpengine?

Any far-end NAT traversal manipulations involved such as SIP ALG or STUN?

 

Cheers.

 

On Thu, Mar 22, 2018 at 3:55 PM, gerry kernan <gerry.kernan@infinityit.ie> wrote:

Hi

 

Hoping someone can point me in the right direction.

I have a Kamailio Ver: 4.2.3-1.1  running in front of a few asterisk servers Ver: 13.17.2  sip is routed to an asterisk server depending the domain name in the sip request, all working as expected . but if a call is put on hold  after resuming the call the party that placed the call on hold can’t hear any audio. The other party can hear . do I need to do anything special to handle re-invites for calls put on hold?

 

 

Gerry Kernan

 

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