Strange rtp error


I have set up a VoIP network with 60 Grandstream GXV 3000 (FW video phones and Asterisk 1.4.11 running on IBM x346 server (3.4 GHz Xeon, 4 Gb RAM, 2 76Gb SCSI HDD’s in RAID 1 array) with Sangoma A101 E1 card. The installation of the software, interface card and configuration of Asterisk all seem to have gone smooth. But I keep getting these errors when I try to establish a call :
NOTICE[30681]: rtp.c:1279 ast_rtp_read: Unknown RTP codec 126received from '
I’m getting 3 of these for both the caller and the callee side. This results in losing either audio or video or sometimes droping of the call. What is the cause of this and how can I get it working like it should?

P.S. I did also do a network trace with Wireshark so I can post it if it’s needed.

THX in advance!

The rtp codec 126 should be the H.264 video codec, try activate the support for this codec adding the line “allow=h264” in the “general” section of sip.conf file.


Marco Bruni

First thx for the fast reply, but the thing is that the codec is enabled in the sip.conf and in phones configuration i can see that it uses rtp payload type of 99 for the H.264… So I’m wondering where does the payload type 126 come from. Wouldn’t it be that if the phone is using 99 for the H.264 and if the codec isn’t enabled in the sip.conf that I would get “unknown rtp codec 99” instead?