erzonden:* Monday, October 18, 2021 11:50:22 PM
*Aan:* OpenSIPS users mailling list
*Onderwerp:* Re: [OpenSIPS-Users] double sdp.
Johan,
To avoid problems like this, I call rtpengine_offer() in branch_route on
initial invites, and make sure to call rtpengine_delete() in any failure
route to r
Johan,
To avoid problems like this, I call rtpengine_offer() in branch_route on
initial invites, and make sure to call rtpengine_delete() in any failure
route to remove any session from a failed offer that was never used.
Perhaps these will help in your situation as well.
- Jeff
On Mon, Oct 18
Hi,
A and B are on the same proxy.
A calls B,
(as I need transcoding I need to call rtpengine_offer here)
B returns 183 with SDP.
(this implies calling rtpengine_answer in onreply_route)
B lets the call time out
On the proxy I intercept the 480 returned by B
and I change the INVITE so t
Thanks!
Now all is OK!
На ср, 4.09.2019 г. в 15:34 ч. Ben Newlin написа:
> If you don't want to have both in the second INVITE, you can try putting
> both rtpengine_offer calls in branch routes instead. I haven't worked with
> rtpengine, but with other messages changes like this if you place the
Hi Dragomir. May be you miss rtpengine_answer on reply?
Like
onreply_route[handle_reply] {
if (has_body("application/sdp"){
$var(rtpengine_flags) = "RTP/AVP replace-session-connection
replace-origin ICE=remove";
}
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Users@lists.op
Hello,
Thanks for your replay.
If that's OK. Why do I have one-way voice?
Rerards,
На ср, 4.09.2019 г. в 15:29 ч. Alexey Vasilyev
написа:
> This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. This
> is
> Invite from snom phone, for example:
>
> Sent to tls:135.42.212.82:5061
If you don't want to have both in the second INVITE, you can try putting both
rtpengine_offer calls in branch routes instead. I haven't worked with
rtpengine, but with other messages changes like this if you place them in the
branch route then they affect only the current branch; after failure t
This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. This is
Invite from snom phone, for example:
Sent to tls:135.42.212.82:5061 at Sep 4 14:19:18.641 (1383 bytes):
INVITE sip:*7...@sip.test.dk SIP/2.0
Via: SIP/2.0/TLS 172.16.1.29:4169;branch=z9hG4bK-gci2vl6fe7cz;rport
From: "Dem
No it is either rtp/avp or rtp/savp.
On Wed, 4 Sep 2019, 13:54 Dragomir Haralambiev, wrote:
> Hello,
>
> I try setup Opensips to make local calls using device with encryption.
>
> In this example, it is not known UA 2 whether it supported (RTP/SAVP).
> UA 1 (RTP/AVP) > Opensips+Rtpengine ---
Hello,
I try setup Opensips to make local calls using device with encryption.
In this example, it is not known UA 2 whether it supported (RTP/SAVP).
UA 1 (RTP/AVP) > Opensips+Rtpengine -> UA 2
1. I try to make call with:
rtpengine_offer("RTP/AVP replace-session-connection replace-origin
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