RTP isn't sent to the negotiated port on RE-Invite

NC
Nuno Centeio
Wed, May 27, 2020 9:48 AM

Hi,

I'm having a problem with PJSIP 2.9 in an Android.
I've successfully built a SIP client but I'm having an issue when I receive
a REINVITE with Replaces headers, where the RTP audio port switches from
the initial invite.

I've attached my log file.

This is where I've received the REINVITE:
10:07:43.167          pjsua_core.c  .RX 1293 bytes Request msg
INVITE/cseq=1 (rdata0x78a5706b30) from TCP X.X.X.X:5050:
INVITE sip:1012@X.X.X.X:5050;transport=TCP;ob SIP/2.0
From: "martinho2"<sip:1010@DomainXPTO

;tag=5bad1b70-b00000a-fd2-65014-2b863-7865b38a-2b863

(...)
Replaces:
f3a2717b-4986-46e7-8a3c-7745f36a6284;to-tag=b003d99b-f717-4d0a-ae50-ad7977bda658;from-tag=5bad10b0-b00000a-fd2-65014-2b863-2cb0c32-2b863
(...)
m=audio 19768 RTP/AVP 111 103 104 9 0 8 106 105 13 110 112 113 126
(...)

In a server side Wireshark I see that my App is sending packets to the port
19714, which is a random port that was never negotiated.
[image: image.png]

This doesn' seem to be a NAT problem as normal calls works perfectly.

Can anyone point me in the right direction?

Thanks

Hi, I'm having a problem with PJSIP 2.9 in an Android. I've successfully built a SIP client but I'm having an issue when I receive a REINVITE with Replaces headers, where the RTP audio port switches from the initial invite. I've attached my log file. This is where I've received the REINVITE: 10:07:43.167 pjsua_core.c .RX 1293 bytes Request msg INVITE/cseq=1 (rdata0x78a5706b30) from TCP X.X.X.X:5050: INVITE sip:1012@X.X.X.X:5050;transport=TCP;ob SIP/2.0 From: "martinho2"<sip:1010@DomainXPTO >;tag=5bad1b70-b00000a-fd2-65014-2b863-7865b38a-2b863 (...) Replaces: f3a2717b-4986-46e7-8a3c-7745f36a6284;to-tag=b003d99b-f717-4d0a-ae50-ad7977bda658;from-tag=5bad10b0-b00000a-fd2-65014-2b863-2cb0c32-2b863 (...) m=audio 19768 RTP/AVP 111 103 104 9 0 8 106 105 13 110 112 113 126 (...) In a server side Wireshark I see that my App is sending packets to the port 19714, which is a random port that was never negotiated. [image: image.png] This doesn' seem to be a NAT problem as normal calls works perfectly. Can anyone point me in the right direction? Thanks
NC
Nuno Centeio
Mon, Jun 22, 2020 6:54 PM

I still have the same problem.
Isn't there anyone who can help me?

Thanks

On Wed, 27 May 2020 at 10:48, Nuno Centeio nuno.r.centeio@gmail.com wrote:

Hi,

I'm having a problem with PJSIP 2.9 in an Android.
I've successfully built a SIP client but I'm having an issue when I
receive a REINVITE with Replaces headers, where the RTP audio port switches
from the initial invite.

I've attached my log file.

This is where I've received the REINVITE:
10:07:43.167          pjsua_core.c  .RX 1293 bytes Request msg
INVITE/cseq=1 (rdata0x78a5706b30) from TCP X.X.X.X:5050:
INVITE sip:1012@X.X.X.X:5050;transport=TCP;ob SIP/2.0
From: "martinho2"<sip:1010@DomainXPTO

;tag=5bad1b70-b00000a-fd2-65014-2b863-7865b38a-2b863

(...)
Replaces:
f3a2717b-4986-46e7-8a3c-7745f36a6284;to-tag=b003d99b-f717-4d0a-ae50-ad7977bda658;from-tag=5bad10b0-b00000a-fd2-65014-2b863-2cb0c32-2b863
(...)
m=audio 19768 RTP/AVP 111 103 104 9 0 8 106 105 13 110 112 113 126
(...)

In a server side Wireshark I see that my App is sending packets to the
port 19714, which is a random port that was never negotiated.
[image: image.png]

This doesn' seem to be a NAT problem as normal calls works perfectly.

Can anyone point me in the right direction?

Thanks

I still have the same problem. Isn't there anyone who can help me? Thanks On Wed, 27 May 2020 at 10:48, Nuno Centeio <nuno.r.centeio@gmail.com> wrote: > Hi, > > I'm having a problem with PJSIP 2.9 in an Android. > I've successfully built a SIP client but I'm having an issue when I > receive a REINVITE with Replaces headers, where the RTP audio port switches > from the initial invite. > > I've attached my log file. > > This is where I've received the REINVITE: > 10:07:43.167 pjsua_core.c .RX 1293 bytes Request msg > INVITE/cseq=1 (rdata0x78a5706b30) from TCP X.X.X.X:5050: > INVITE sip:1012@X.X.X.X:5050;transport=TCP;ob SIP/2.0 > From: "martinho2"<sip:1010@DomainXPTO > >;tag=5bad1b70-b00000a-fd2-65014-2b863-7865b38a-2b863 > (...) > Replaces: > f3a2717b-4986-46e7-8a3c-7745f36a6284;to-tag=b003d99b-f717-4d0a-ae50-ad7977bda658;from-tag=5bad10b0-b00000a-fd2-65014-2b863-2cb0c32-2b863 > (...) > m=audio 19768 RTP/AVP 111 103 104 9 0 8 106 105 13 110 112 113 126 > (...) > > In a server side Wireshark I see that my App is sending packets to the > port 19714, which is a random port that was never negotiated. > [image: image.png] > > This doesn' seem to be a NAT problem as normal calls works perfectly. > > Can anyone point me in the right direction? > > Thanks > > >