No rtp activity while in conversation

Asterisk Asterisk SIP
1

Hello all,

While i am on a call, having a conversation (Rtp flow is all fine on rtp debug), the call drops. The strange part is that the pattern is not always reproducable. Sometimes, when i make a call to another destination (which also goes out through the same nic and to the same ip), call stays on and no drop.

What could be it ? When i look at the SIP trace, i see a BYE message initiated from the asterisk, same time as i see “lack of rtp activity” message on the asterisk log. Note, this setup has been running well for over few years now. No changes were made whatsoever.

Sent RTP packet to 1.2.3.4:15436 (type 08, seq 030069, ts 3406293600, len 000160)
Got RTP packet from 5.6.7.8:12474 (type 00, seq 002108, ts 3406293766, len 000160)
Sent RTP packet to 1.2.3.4:15436 (type 08, seq 030070, ts 3406293760, len 000160)
Got RTP packet from 5.6.7.8:12474 (type 00, seq 002109, ts 3406293926, len 000160)

chan_sip.c:29991 check_rtp_timeout: Disconnecting call ‘SIP/sip-00000001’ for lack of RTP activity in 6 seconds

2

Without the timestamps, that log is not really useful. The timestamps should be present in the full log file, which may need enabling.

At the moment, I’d have to take this at face value, and say that you have stopped receiving RTP.

3

Here are the “rtp set debug on” output and the “sip set debug on” . I don;t know if this is readable. Let me know please. This call dropped while in conversation for about 40 plus seconds. A BYE is sent first from the asterisk. I could not paste the entire debug output here due to character limitation. So, i have uploaded it in a google drive and the url is below:

Please help me out.

120.56.42.94 is my switch ip (sending calls to asterisk)
103.94.21.96 is my internet router in front of the asterisk
172.23.201.98 is my asterisk nic 2 interface ip. This is connected to the sip networ
172.24.23.2 is the SIP signaling ip
172.24.22.116 is the media ip

4

It says “This Doc is private”. However, the router IP should not be visible in the logs. If it is, you need to find out why. You may need to disable SIP ALG on the router, as that is normally broken.

Also note that if media and signalling address are those configured in Asterisk, you need to explain why they are different. You may have configured those from the wrong end.

5

Try the google doc link once. I have made it available now.

The SIP ALG on the router is not enabled. On the asterisk sip.conf, we see only the signaling ip address configured. the media ip (that is what the sip provider said) appears on the rtp stream only.

[sip]

host=172.24.23.2
qualify=yes
type=friend
port=5060
context=incoming
nat=force_rport,comedia
call-limit=16
disallow=all
allow=ulaw
allow=alaw
allow=g729
insecure=port,invite
dtmfmode=rfc2833

6

@david551 - Any suggestion?

7

After this point:

Got  RTP packet from    120.56.42.94:34574 (type 00, seq 002152, ts 2501871651, len 000160)
Sent RTP packet to      172.24.22.116:13614 (type 08, seq 011408, ts 2501871648, len 000160)
Got  RTP packet from    172.24.22.116:13614 (type 08, seq 050461, ts 419302552, len 000160)
Sent RTP packet to      120.56.42.94:34574 (type 00, seq 013091, ts 419302552, len 000160)
Got  RTP packet from    120.56.42.94:34574 (type 00, seq 002153, ts 2501871811, len 000160)
Sent RTP packet to      172.24.22.116:13614 (type 08, seq 011409, ts 2501871808, len 000160)
Got  RTP packet from    120.56.42.94:34574 (type 00, seq 002154, ts 2501871971, len 000160)
Sent RTP packet to      172.24.22.116:13614 (type 08, seq 011410, ts 2501871968, len 000160)
Got  RTP packet from    120.56.42.94:34574 (type 00, seq 002155, ts 2501872131, len 000160)
Got  RTP packet from    120.56.42.94:34574 (type 00, seq 002156, ts 2501872291, len 000160)
Sent RTP packet to      172.24.22.116:13614 (type 08, seq 011411, ts 2501872128, len 000160)
Sent RTP packet to      172.24.22.116:13614 (type 08, seq 011412, ts 2501872288, len 000160)

you stopped receiving from 172.24.22.116, which appears to be the B side.

Unfortunately there are no timestamps, probably because you screen scraped, instead of using log files, but I think this is a long way from when Asterisk did anything significant, so I’d say the problem was with 172.24.22.116 or the network. You need to capture RTP up stream of Asterisk and see if it is being sent.

It is also possible that 172.24.22.116 has tried to do a SIP exchange, and that hasn’t got through, and it has aborted the call as a result, but because SIP isn’t getting through, the only evidence is the lack of RTP. Generally you need to capture near 172.24.22.116.

It looks to me as though there is significant jitter on the A side.

8

Strange behavior is - if we eliminate asterisk and connect my pc direct the B-side, then there is no isssue . Call doesn’t drop and lasts as long as it could until it is hung up.

You said there is jutter on the A side ? How did you notice that ? How can we fix it.?

9

Could it be a bad network interface card? Possible?

10

Rtp packet to from should be same address

On Fri, 25 Oct 2024, 20:39 muslimak via Asterisk Community, <notifications@asterisk.discoursemail.com> wrote:

muslimak
October 25

Could it be a bad network interface card? Possible?

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11

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