]" in my log about rtpengine HOT 11 CLOSED

tektak commented on August 24, 2024

Hi, I am new to kamailio and rtpengine.. I am trying to implement example as per https://github.com/caruizdiaz/kamailio-ws. The call is plased successfully btw. http://tryit.jssip.net/ and softfone but no audio video is received. I get "ERROR:

Comments (11)

tektak commented on August 24, 2024

The log grom RTPEngine is
Oct 14 13:07:32 TN02028 rtpengine[3725]: Got valid command from 127.0.0.1:35124: ping - { "command": "ping" }
Oct 14 13:07:32 TN02028 rtpengine[3725]: Returning to SIP proxy: d6:result4:ponge
Oct 14 13:07:32 TN02028 rtpengine[3725]: Got valid command from 127.0.0.1:35124: ping - { "command": "ping" }
Oct 14 13:07:32 TN02028 rtpengine[3725]: Returning to SIP proxy: d6:result4:ponge
Oct 14 13:09:01 TN02028 CRON[5310]: (root) CMD ( [ -x /usr/lib/php5/maxlifetime ] && [ -x /usr/lib/php5/sessionclean ] && [ -d /var/lib/php5 ] && /usr/lib/php5/sessio$
Oct 14 13:09:01 TN02028 CRON[5310]: (root) CMD ( [ -x /usr/lib/php5/maxlifetime ] && [ -x /usr/lib/php5/sessionclean ] && [ -d /var/lib/php5 ] && /usr/lib/php5/sessio$
Oct 14 13:09:16 TN02028 rtpengine[3725]: Got valid command from 127.0.0.1:39976: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-32.0.3 263 0 IN IP4 0.0.0.0#015#012s=SIP $
Oct 14 13:09:16 TN02028 rtpengine[3725]: Got valid command from 127.0.0.1:39976: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-32.0.3 263 0 IN IP4 0.0.0.0#015#012s=SIP $
Oct 14 13:09:16 TN02028 rtpengine[3725]: ... -15#015#012a=sendrecv#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=candid$
Oct 14 13:09:16 TN02028 rtpengine[3725]: Unknown flag encountered: 'force'
Oct 14 13:09:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Creating new call
Oct 14 13:09:16 TN02028 rtpengine[3725]: ... -15#015#012a=sendrecv#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=candid$
Oct 14 13:09:16 TN02028 rtpengine[3725]: Unknown flag encountered: 'force'
Oct 14 13:09:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Creating new call
Oct 14 13:09:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Returning to SIP proxy: d3:sdp301:v=0#015#012o=Mozilla-SIPUA-32.0.3 263 0 IN IP4 0.0.0.0#015#012s=SIP C$
Oct 14 13:09:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Returning to SIP proxy: d3:sdp301:v=0#015#012o=Mozilla-SIPUA-32.0.3 263 0 IN IP4 0.0.0.0#015#012s=SIP C$
Oct 14 13:09:24 TN02028 rtpengine[3725]: Got valid command from 127.0.0.1:48350: answer - { "sdp": "v=0#015#012o=doubango 1983 678901 IN IP4 10.0.4.109#015#012s=-#15#$
Oct 14 13:09:24 TN02028 rtpengine[3725]: Got valid command from 127.0.0.1:48350: answer - { "sdp": "v=0#015#012o=doubango 1983 678901 IN IP4 10.0.4.109#015#012s=-#15#$
Oct 14 13:09:24 TN02028 rtpengine[3725]: ... ", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/SAVPF"$
Oct 14 13:09:24 TN02028 rtpengine[3725]: Unknown flag encountered: 'force'
Oct 14 13:09:24 TN02028 rtpengine[3725]: ... ", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/SAVPF"$
Oct 14 13:09:24 TN02028 rtpengine[3725]: Unknown flag encountered: 'force'
Oct 14 13:09:24 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Returning to SIP proxy: d3:sdp769:v=0#015#012o=doubango 1983 678901 IN IP4 192.168.2.103#015#012s=-#15$
Oct 14 13:09:24 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Returning to SIP proxy: d3:sdp769:v=0#015#012o=doubango 1983 678901 IN IP4 192.168.2.103#015#012s=-#15$
Oct 14 13:09:24 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] ... :active#015#012a=fingerprint:sha-1 DF:55:74:F2:D6:7E:AD:88:8D:10:7A:D0:10:4D:10:CA:8B:88:46:75#015#$
Oct 14 13:09:24 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] ... :active#015#012a=fingerprint:sha-1 DF:55:74:F2:D6:7E:AD:88:8D:10:7A:D0:10:4D:10:CA:8B:88:46:75#015#$
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Closing call due to timeout
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Closing call due to timeout
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Final packet stats:
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] --- Tag '838828592', created 1:00 ago, in dialogue with 'bcm45c6mss'
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] ------ Media #1, port 30424 <> 10.0.4.109:16898, 0 p, 0 b, 0 e
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] Final packet stats:
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] --- Tag '838828592', created 1:00 ago, in dialogue with 'bcm45c6mss'
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] ------ Media #1, port 30424 <> 10.0.4.109:16898, 0 p, 0 b, 0 e
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] --- Tag 'bcm45c6mss', created 1:00 ago, in dialogue with '838828592'
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] ------ Media #1, port 30434 <> 10.0.2.120:58279, 0 p, 0 b, 0 e
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] --- Tag 'bcm45c6mss', created 1:00 ago, in dialogue with '838828592'
Oct 14 13:10:16 TN02028 rtpengine[3725]: [p5hmbg7l5fn7c2h0uea7] ------ Media #1, port 30434 <> 10.0.2.120:58279, 0 p, 0 b, 0 e
Oct 14 13:17:01 TN02028 CRON[5345]: (root) CMD ( cd / && run-parts --report /etc/cron.hourly)
Oct 14 13:17:01 TN02028 CRON[5345]: (root) CMD ( cd / && run-parts --report /etc/cron.hourly)

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tektak commented on August 24, 2024

And the Browser console shows
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS bbggv6665uc4.invalid;branch=z9hG4bK7917522
Max-Forwards: 69
To: sip:[email protected]
From: "1060" sip:[email protected];tag=bcm45c6mss
Call-ID: p5hmbg7l5fn7c2h0uea7
CSeq: 9638 INVITE
Contact: sip:[email protected];transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 660

v=0
o=Mozilla-SIPUA-32.0.3 263 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:99731384
a=ice-pwd:5fbcfc25a365da28ed3c03fc43b331aa
a=fingerprint:sha-256 6D:1F:6C:89:9F:1F:2D:38:54:A9:63:73:FA:19:21:04:8D:3B:BE:19:0D:8C:94:D4:8C:91:EE:F7:4A:99:A2:4A
m=audio 58279 RTP/SAVPF 109 0 8 101
c=IN IP4 10.0.2.120
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=candidate:0 1 UDP 2128609535 10.0.2.120 58279 typ host
a=candidate:0 2 UDP 2128609534 10.0.2.120 58280 typ host
a=rtcp-mux

jssip-devel.js (line 519)

JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/WS bbggv6665uc4.invalid;branch=z9hG4bK7917522;rport=12796;received=10.0.2.120
To: sip:[email protected]
From: "1060" sip:[email protected];tag=bcm45c6mss
Call-ID: p5hmbg7l5fn7c2h0uea7
CSeq: 9638 INVITE
Server: kamailio (4.3.0-dev0 (x86_64/linux))
Content-Length: 0

jssip-devel.js (line 686)

JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 180 Ringing
From: "1060"sip:[email protected];tag=bcm45c6mss
To: sip:[email protected];tag=838828592
Contact: sip:[email protected]:49156;transport=udp
Call-ID: p5hmbg7l5fn7c2h0uea7
CSeq: 9638 INVITE
Content-Length: 0
Via: SIP/2.0/WS bbggv6665uc4.invalid;rport=12796;received=10.0.2.120;branch=z9hG4bK7917522
Record-Route: sip:10.0.2.28:5060;nat=yes;lr=on
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

jssip-devel.js (line 686)
JsSIP | DIALOG | new UAC dialog created with status EARLY
jssip-devel.js (line 2546)
JsSIP | EVENT EMITTER | emitting event progress
jssip-devel.js (line 187)

JsSIP | TRANSPORT | received WebSocket text message:

jssip-devel.js (line 686)
JsSIP | EVENT EMITTER | emitting event progress
jssip-devel.js (line 187)

JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
From: "1060"sip:[email protected];tag=bcm45c6mss
To: sip:[email protected];tag=838828592
Contact: sip:[email protected]:49156;transport=udp
Call-ID: p5hmbg7l5fn7c2h0uea7
CSeq: 9638 INVITE
Content-Type: application/sdp
Content-Length: 769
Via: SIP/2.0/WS bbggv6665uc4.invalid;rport=12796;received=10.0.2.120;branch=z9hG4bK7917522
Record-Route: sip:10.0.2.28:5060;nat=yes;lr=on
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=doubango 1983 678901 IN IP4 192.168.2.103
s=-
c=IN IP4 192.168.2.103
t=0 0
a=ice-lite
m=audio 30434 RTP/SAVPF 0 8 101
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ssrc:1315460377 cname:doubango@1746941213
a=ssrc:1315460377 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1315460377 label:doubango@audio
a=sendrecv
a=rtcp:30434
a=rtcp-mux
a=setup:active
a=fingerprint:sha-1 DF:55:74:F2:D6:7E:AD:88:8D:10:7A:D0:10:4D:10:CA:8B:88:46:75
a=ice-ufrag:pctjzzKV
a=ice-pwd:OInKlCPQ784G7UpzX9WS6Y6t9zMI
a=candidate:bjdPSFzrFqPpoLxM 1 UDP 2130706431 192.168.2.103 30434 typ host
a=candidate:U1vWeD8KW2seuxvF 1 UDP 2130706175 186.0.191.55 30434 typ host

jssip-devel.js (line 686)

JsSIP | DIALOG | dialog p5hmbg7l5fn7c2h0uea7bcm45c6mss838828592 changed to CONFIRMED state

jssip-devel.js (line 2557)

JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:[email protected]:49156;transport=udp SIP/2.0
Route: sip:10.0.2.28:5060;nat=yes;lr=on
Via: SIP/2.0/WS bbggv6665uc4.invalid;branch=z9hG4bK3516541
Max-Forwards: 69
To: sip:[email protected];tag=838828592
From: "1060" sip:[email protected];tag=bcm45c6mss
Call-ID: p5hmbg7l5fn7c2h0uea7
CSeq: 9638 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

Can anybody help me solve this problem,
Thanks!!!

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rfuchs commented on August 24, 2024

Your posted log output is incomplete, the lines are truncated. At first glance though, it appears that rtpengine isn't actually receiving any media packets at all. Perhaps you're trying to run it from behind NAT?

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tektak commented on August 24, 2024

Hi!!! rfuchs, thanks for the reply. as you said to run without NAT, when i run without NAT in https://github.com/caruizdiaz/kamailio-ws/blob/master/kamailio-ws.cfg, i am not able to connect to kamailio and get other errors. also u said my logs are incomplete so, I am attaching my complete log WITH_NAT.

rtpengine: http://pastebin.com/TfeqBnc0
Kamailio: https://drive.google.com/file/d/0B5c8y3uuaFcZX3o2WU1rTnpPSWs/
Browser: http://pastebin.com/fvyQSWjv
Please help me.
Thank You!!

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rfuchs commented on August 24, 2024

What I mean is that your SIP endpoints seem to be 10.0.2.120 and 10.0.4.109, but rtpengine seems to be 192.168.2.103. Can these different networks communicate with each other directly?

The final packet report says:

Oct 15 11:03:11 TN02028 rtpengine[3975]: [q19kdh6il1bjrdu7bnsp] ------ Media #1, port 30000 <>      10.0.4.109:45254, 0 p, 0 b, 0 e
Oct 15 11:03:11 TN02028 rtpengine[3975]: [q19kdh6il1bjrdu7bnsp] ------ Media #1, port 30010 <>      10.0.2.120:54391, 0 p, 0 b, 0 e

Which indicates that nothing was ever received on any of the ports (all counters are zero). I don't see anything wrong in the logs, so I'm guessing that you have a network problem somewhere.

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tektak commented on August 24, 2024

Thanks Richard, there was network problem.. I had provided wrong IP for rtpengine run... IT's working fine now.. Thank You very much.

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tektak commented on August 24, 2024

Hi Richard i got one more problem, the mentioned config works fine for call between softphone and web but i get error while placing call between web to web it says " SRTP output wanted, but no crypto suite was negotiated". here is the log from rtpengine:http://pastebin.com/0Gztmbgn

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rfuchs commented on August 24, 2024

This is somewhat expected in both of the calls in your log. It's primarily a client problem.

In the first call, you're proxying SRTP to SRTP. Side A sends "offer", which is relayed to side B. Rtpengine initiates DTLS to side A, while side B initiates DTLS to rtpengine. Side B complete DTLS handshake and starts sending SRTP. However, side A hasn't responded to the DTLS handshake yet and so rtpengine cannot forward the SRTP packets. After several seconds, side B finally sends the "answer", which is relayed to side A, at which point side A finally takes up on the DTLS handshake. After this, SRTP can flow. You can confirm this by running a packet dump and I believe this is a known problem in Firefox.

A similar situation exists for the second call, which proxies SRTP to RTP. Side B (RTP) starts sending RTP before the DTLS handshake on side A could be completed. Again I suspect that Firefox isn't responding to the DTLS packets until several seconds after the "answer" has been received. Only after this can SRTP packets flow. Again you can confirm this by doing a packet dump.

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tektak commented on August 24, 2024

For "WEB to WEB call" I get problem with chrome too,, http://tryit.jssip.net/ works fine both with chrome and Firefox, but http://sipml5.org/call.htm doesn't work with neither.
Also http://sipml5.org/call.htm works well for "Web to Softphone" and "Softphone to Web".
The log for chrome is:http://pastebin.com/wncLJTuK
and log from kamailio is :http://pastebin.com/43cmYTPn

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rfuchs commented on August 24, 2024

In this case, there only was an "offer" from Chrome but no "answer" from the called endpoint. Also, no packets at all were received, indicating a possible network issue. Entirely different problems here.

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tektak commented on August 24, 2024

Hi! now i am able to make call but no audio video. Every thing is working fine except calls btw. web with http://sipml5.org/call.htm. i don't know whether its problem in kamailio's config, rtpengine or network. if the problem is in network what might be the problem.. everything is working fine. i also disabled ipv6.
here are my complete log
http://pastebin.com/gg68dqeN
http://pastebin.com/uabndg5h

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