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RTP Timeout

Added by Jorge Cornejo 3 months ago

Hi
I am getting an RTP timeout at agent interface. As for what I understand this is because Kamailio can't connect to the rtpengine?
I can't find a proper call flow sample in order to understand exactly what is supposed to happen when communicating between components.
So what I understand is client webrtc connects to Kamailio
Kamailio connects to Asterisk
Asterisk connects to rtpengine
Is this correct?

INVITE sip:2347600834@subdomain.maindomain.com SIP/2.0
Via: SIP/2.0/UDP 172.31.14.56:5070;branch=z9hG4bK249bdd38;rport
Max-Forwards: 70
From: "S1903140207028600051" <sip:5164536886@172.31.14.56:5070>;tag=as4c75660f
To: <sip:2347600834@subdomain.maindomain.com>
Contact: <sip:5164536886@172.31.14.56:5070>
Call-ID: 767e2d6446c81f155ab07c037e8e705f@172.31.14.56:5070
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.17.2-vici
Date: Wed, 13 Mar 2019 18:07:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Remote-Party-ID: "S1903140207028600051" <sip:5164536886@172.31.14.56>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 1694008216 1694008216 IN IP4 172.31.14.56
s=Asterisk PBX 13.17.2-vici
c=IN IP4 172.31.14.56
t=0 0
m=audio 18816 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 172.31.14.56:5070;branch=z9hG4bK249bdd38;rport=5070;received=182.128.207.214
From: "S1903140207028600051" <sip:5164536886@172.31.14.56:5070>;tag=as4c75660f
To: <sip:2347600834@subdomain.maindomain.com>
Call-ID: 767e2d6446c81f155ab07c037e8e705f@172.31.14.56:5070
CSeq: 102 INVITE
Content-Length: 0

SIP/2.0 180 Ringing
Record-Route: <sip:172.31.14.56:4443;transport=ws;r2=on;lr;nat=yes>
Record-Route: <sip:182.128.207.214;r2=on;lr;nat=yes>
Via: SIP/2.0/UDP 172.31.14.56:5070;received=182.128.207.214;branch=z9hG4bK249bdd38;rport=5070
To: <sip:2347600834@subdomain.maindomain.com>;tag=eob52d21ag
From: "S1903140207028600051" <sip:5164536886@172.31.14.56:5070>;tag=as4c75660f
Call-ID: 767e2d6446c81f155ab07c037e8e705f@172.31.14.56:5070
CSeq: 102 INVITE
Contact: <sip:m46dnpij@ge825hepi2ih.invalid;transport=ws;alias=189.6.241.13~4837~6;alias=189.6.241.13~4837~6>
Supported: ice,replaces,outbound
Content-Length: 0

SIP/2.0 200 OK
Record-Route: <sip:172.31.14.56:4443;transport=ws;r2=on;lr;nat=yes>
Record-Route: <sip:182.128.207.214;r2=on;lr;nat=yes>
Via: SIP/2.0/UDP 172.31.14.56:5070;received=182.128.207.214;branch=z9hG4bK249bdd38;rport=5070
To: <sip:2347600834@subdomain.maindomain.com>;tag=eob52d21ag
From: "S1903140207028600051" <sip:5164536886@172.31.14.56:5070>;tag=as4c75660f
Call-ID: 767e2d6446c81f155ab07c037e8e705f@172.31.14.56:5070
CSeq: 102 INVITE
Contact: <sip:m46dnpij@ge825hepi2ih.invalid;transport=ws;alias=189.6.241.13~4837~6;alias=189.6.241.13~4837~6>
Supported: ice,replaces,outbound
Content-Type: application/sdp
Content-Length: 686
Session-Expires: 90;refresher=uac

v=0
o=- 83891966193305818 2 IN IP4 172.31.14.56
s=-
t=0 0
a=msid-semantic: WMS WZlHxSfyEEfSz3aMHXiWw7bcJQyqTNj0PDM9
m=audio 30012 RTP/AVP 107 0 101
c=IN IP4 172.31.14.56
a=msid:WZlHxSfyEEfSz3aMHXiWw7bcJQyqTNj0PDM9 130d1922-e789-427c-a229-88f21f530d07
a=ssrc:3277383668 cname:cvvQJ123nfOVozWV
a=ssrc:3277383668 msid:WZlHxSfyEEfSz3aMHXiWw7bcJQyqTNj0PDM9 130d1922-e789-427c-a229-88f21f530d07
a=ssrc:3277383668 mslabel:WZlHxSfyEEfSz3aMHXiWw7bcJQyqTNj0PDM9
a=ssrc:3277383668 label:130d1922-e789-427c-a229-88f21f530d07
a=rtpmap:107 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:107 minptime=10;useinbandfec=1
a=sendrecv
a=rtcp:30013
a=ptime:20
ACK sip:m46dnpij@ge825hepi2ih.invalid;transport=ws;alias=189.6.241.13~4837~6;alias=189.6.241.13~4837~6 SIP/2.0
Via: SIP/2.0/UDP 172.31.14.56:5070;branch=z9hG4bK6b5931fc;rport
Route: <sip:182.128.207.214;r2=on;lr;nat=yes>,<sip:172.31.14.56:4443;transport=ws;r2=on;lr;nat=yes>
Max-Forwards: 70
From: "S1903140207028600051" <sip:5164536886@172.31.14.56:5070>;tag=as4c75660f
To: <sip:2347600834@subdomain.maindomain.com>;tag=eob52d21ag
Contact: <sip:5164536886@172.31.14.56:5070>
Call-ID: 767e2d6446c81f155ab07c037e8e705f@172.31.14.56:5070
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.17.2-vici
Content-Length: 0

BYE sip:5164536886@172.31.14.56:5070 SIP/2.0
Via: SIP/2.0/UDP 182.128.207.214:5060;branch=z9hG4bK2a.45fea5a9c3f4e04b603a3ac447c11a69.0
Via: SIP/2.0/WSS ge825hepi2ih.invalid;rport=4837;received=189.6.241.13;branch=z9hG4bK7280609
Max-Forwards: 68
To: <sip:5164536886@172.31.14.56:5070>;tag=as4c75660f
From: <sip:2347600834@subdomain.maindomain.com>;tag=eob52d21ag
Call-ID: 767e2d6446c81f155ab07c037e8e705f@172.31.14.56:5070
CSeq: 4345 BYE
Reason: SIP ;cause=408; text="RTP Timeout" 
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.13
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 182.128.207.214:5060;branch=z9hG4bK2a.45fea5a9c3f4e04b603a3ac447c11a69.0;received=182.128.207.214;rport=5060
Via: SIP/2.0/WSS ge825hepi2ih.invalid;rport=4837;received=189.6.241.13;branch=z9hG4bK7280609
From: <sip:2347600834@subdomain.maindomain.com>;tag=eob52d21ag
To: <sip:5164536886@172.31.14.56:5070>;tag=as4c75660f
Call-ID: 767e2d6446c81f155ab07c037e8e705f@172.31.14.56:5070
CSeq: 4345 BYE
Server: Asterisk PBX 13.17.2-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


Replies (6)

RE: RTP Timeout - Added by Jorge Cornejo 3 months ago

BTW I am using an EC2 instance as front, so NAT is going on.
Thanks

RE: RTP Timeout - Added by Jorge Cornejo 3 months ago

182.128.207.214 EC2 Public IP
172.31.14.56 EC2 LAN IP
189.6.241.13 My PC Public IP
192.168.0.2 My PC LAN IP

RE: RTP Timeout - Added by Jorge Cornejo 3 months ago

Changing nat=yes at asterisk configuration file solved the issue. Still trying to make a test call.

RE: RTP Timeout - Added by Jorge Cornejo 3 months ago

Seems it didn't work...I think I have created another scenario.
Will validate all over again.

RE: RTP Timeout - Added by Demian Lizandro Biscocho 3 months ago

Were you able to make it work with Amazon EC2?

RE: RTP Timeout - Added by Jorge Cornejo 3 months ago

No...something is missing between Kamailio and Asterisk...I am about to drop it off :(

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