Asterisk 13.8.2 res_rtp_asterisk PJ ICE ERROR 370400 bad request

I’m using asterisk 13.8.2 and want to make a end to end ipphone video/audio and RTT realtime text.
I’m using NAT and STUN so it requires the PJPROJECT BUILD. I set up my stun server address and port in sip.conf as well as rtp.con and res_stun_monitor.conf
I’m also setting proper configuration for t140 and red.

I used configure with pjproject_bundle to build

I’m able to make a video/audio call but text failed.

On PCAP I’m getting the STUN error : res_rtp_asterisk.c: PJ ICE Rx error status code: 370400 ‘Bad Request’
mainly on UDP text port so perhaps that’s why RTT failed.

I see in the pcap file that error has something to do with pjnath message.

Please let me know if I have issue with asterisk configuration or with the build that caused the error and text to fail. Does this version of Asterisk support RTT through NAT/STUN properly?

Thank you

Hi -

also noted from above that i’m using my own stun server that works fine with other asterisk releases.

I have to use sip.conf for this asterisk because it supports RTT, The pjsip.conf does not support RTT

I do need the pjproject_bundled so I can use STUN/NAT. correct?

ICE/STUN/TURN has not been really tested with text, so it’s unknown what will happen with it.

Thank you for getting back! Do you imply that it’s not supported? Do you know what releases of pjproject/asterisk have been supported for STUN with RTT?

RTT is not commonly used, and I don’t think it has been tested on any version.

Thank you. Do you know if RTT works in this version without using STUN/NAT? or is it not confirmed that it supports at all?

I don’t know. The feature really really really isn’t used much, if at all. There has been one person in the last few weeks submitting fixes for it, but that’s all I know of.

I see. would you happen to have the topic id or name that you could forward to me?
What was the fix about and for what release of asterisk? Do you have a plan to support RTT T.140 in pjsip at all anytime soon? I know currently RTT is not supported.

Searching Gerrit for “text” will find them. As for supporting It in PJSIP, I know of noone working on such a thing.


I can’t find any topic with Gerrit and text. Could you please help locate it? I searched on this asterisk forum search and can’t find it.

All code that goes into Asterisk is done through code review so they can be searched on there[1].


Could you please let me know what version of Asterisk and pjproject support RTT T.140? I know you mentioned that it has never been tested with the releases before but appreciate if you would know what versions support it. or if you could point to the documentation about the support for the release of RTT.
I’m planning to use pjproject 2.7.2 and asterisk 13 LTS cert.
Are these versions support it?


No version of chan_pjsip supports it. It has not been tested with chan_sip when used with ICE/STUN/TURN.

Thanks. Do you mean it’s not supported on chan_sip at all?
I meant the RTT in chan_sip (sip.conf) and not pjsip.conf

It said on asterisk version 13LTS. that seems like it supports it. could you please confirm?

Text Messaging

  • Asterisk now has protocol independent support for processing text messages
    outside of a call. Messages are routed through the Asterisk dialplan.
    SIP MESSAGE and XMPP are currently supported. There are options in
    jabber.conf and sip.conf to allow enabling these features.
    -> jabber.conf: see the “sendtodialplan” and “context” options.
    -> sip.conf: see the “accept_outofcall_message”, “auth_message_requests”
    and “outofcall_message_context” options.
    The MESSAGE() dialplan function and MessageSend() application have been
    added to go along with this functionality. More detailed usage information
    can be found on the Asterisk wiki (
  • If real-time text support (T.140) is negotiated, it will be preferred for
    sending text via the SendText application. For example, via SIP, messages
    that were once sent via the SIP MESSAGE request would be sent via RTP if
    T.140 text is negotiated for a call.

It has not been tested with ICE/STUN/TURN in chan_sip, and it is rarely (if ever) used by people and has no tests. It may or may not work.

Thanks. Would it work without STUN/TURN/ICE do you know?

I honestly have no idea. I’ve never personally used it. Per my earlier comment someone has been doing fixes, so it may not have worked and they are trying to make it work now. When I say it is rarely used I mean it - I’ve never heard of anyone using it.

Hello. could you please let me know which version of pjproject supports for asterisk 13.8.2? I am trying pjproject 2.5.5 with asterisk 13.8.2 and asterisk does not start.

is there documentation somewhere indicating which version of pjproject support with the version of asterisk?


There isn’t documentation specifically for that, but using a version from that time should work. New versions can change things, resulting in them not building.

I got this error sending from the phone to the asterisk in pcap file. something about pjnath 2.5.5 svn bad request

Binding error: response error code 400 (bad request);

Does this mean that the pj project does not support nat properly for Realtime text or something is wrong with the build?