Asterisk Is Dropping RTP Packets From Twilio SIP Trunk

Asterisk Asterisk SIP
1

aspire03-desktop*CLI> channel originate PJSIP/+918401345429@twilio-out extension 1000@phones
– Called +918401345429@twilio-out

0x73a2b0007790 – Strict RTP learning after remote address set to: 168.86.139.233:12764
– PJSIP/twilio-out-00000001 is making progress
[Sep 18 15:45:06] WARNING[7858]: pjproject: <?>: sip_transport.c Dropping 172 bytes packet from UDP 168.86.139.233:12764 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 1

[Sep 18 15:45:06] WARNING[7858]: pjproject: <?>: sip_transport.c Dropping 172 bytes packet from UDP 168.86.139.233:12764 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 1

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That would make it seem as though the packets are showing up at the SIP port… and not the RTP port.

Are you behind a NAT? Have you forwarded the external RTP ports to the SIP port?

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Yes i am behind the NAT. i have firewall. but just in my local machine not in public ip. so i am getting the RTP packets from public ip but asterisk is rejecting i dont know why.

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I’ve stated why. They are being sent to the SIP port, not the RTP port. You need to determine why. I’ve stated a possible reason, but it’s up to you to investigate. I am fairly certain this is not an Asterisk issue.

5

In Incoming Calls this is Working perfect but for outgoing that is not working proper. i have check and debug that. can you help me to solved this problem ?

6

You’ll need to look at the actual path of traffic, get a packet capture and see if it is going to the SIP port instead of the RTP port. Look at the SDP and see if a different RTP port was given by Asterisk. Maybe it’s your router screwing up. If Asterisk sends the correct local RTP port in the SDP, but the remote side doesn’t send the RTP there - then something outside of Asterisk is messing up and you have to figure out what.

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