5 thoughts on - Asterisk 13.1.0/PJSIP Peer IP Address Issue

• It would appear that you have the Asterisk server on a public IP address, your two endpoints are behind a NAT, and you have rewrite_contact enabled in pjsip.conf.

  In which case, what you are seeing is correct. In order to be able to send a call to an extension where it is behind NAT, Asterisk must update the contact to have the current IP and port that the phone registered via (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state for).

• Thank you for your note, Scott.

  I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information)
  using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now

  [transport-udp]
  type=transport protocol=udp bind=0.0.0.0
  local_net=192.168.1.0/24
  ;Templates for the necessary config sections

  [endpoint_internal](!)
  type=endpoint context=from-internal disallow=all allow=ulaw

  [auth_userpass](!)
  type=auth auth_type=userpass

  [aor_dynamic](!)
  type=aor max_contacts=1
  remove_existing=yes
  ;Definitions for our phones, using the templates above

  [demo-alice](endpoint_internal)
  auth=demo-alice aors=demo-alice mailboxes=box_a rewrite_contact=yes
  [demo-alice](auth_userpass)
  password=demo-alice ; put a strong, unique password here instead username=demo-alice

  [demo-alice](aor_dynamic)

  [demo-bob](endpoint_internal)
  auth=demo-bob aors=demo-bob mailboxes=box_b rewrite_contact=yes
  [demo-bob](auth_userpass)
  password=demo-bob ; put a strong, unique password here instead username=demo-bob

  [demo-bob](aor_dynamic)

  Thank you for your help!

• Well, I thought it worked, but it actually doesn’t–I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24 network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is appreciated.

  Does the fact that Asterisk is running on a VirtualBox VM on the same machine as one of the SIP phones matter? I am able to access the ARI REST
  interface of the Asterisk server quite fine on the host machine.

  I suspect it has to do with RTP not being set up, but all the codec support is there. Here’s a log for the SIP request from 192.168.1.50:

  <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 --->
  INVITE sip:6002@192.168.1.139;transport=UDP SIP/2.0
  Via: SIP/2.0/UDP 146.115.163.234:64009
  ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1—d8754z-
  Max-Forwards: 70
  Contact:
  To:
  From: ;tag=b661670b Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. CSeq: 2 INVITE
  Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.21933 r21903

  Authorization: Digest username=”demo-alice”,realm=”asterisk”,nonce=”[removed]”,uri=”
  sip:6002@192.168.1.139
  ;transport=UDP”,response=”[removed]”,cnonce=”[removed]”,nc=00000001,qop=auth,algorithm=md5,opaque=”[removed]”

  Allow-Events: presence, kpml Content-Length: 245

  v=0
  o=Z 0 0 IN IP4 146.115.163.234
  s=Z
  c=IN IP4 146.115.163.234
  t=0 0
  m=audio 8000 RTP/AVP 0 3 110 8 98 101
  a=rtpmap:110 speex/8000
  a=rtpmap:98 iLBC/8000
  a=fmtp:98 mode=20
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15
  a=sendrecv

  <--- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 --->
  SIP/2.0 100 Trying Via: SIP/2.0/UDP 146.115.163.234:64009
  ;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1—d8754z-
  Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. From: ;tag=b661670b To:
  CSeq: 2 INVITE
  Content-Length: 0

  Any help is appreciated. A topology is shown below in ASCII.

  < ( Big bad Internet ) >

  GW/NAPT/Router
  |
  ——————————————————–

• Solved!

  The issue was that RTP flows were being established to the wrong IP address.

  I figured out this issue–I had to disable STUN in both SIP phones for this to work correctly.

  Still, I wish a working configuration for Asterisk, and two SIP phones in the same 192.168.1.0/24 network would have helped me tremendously.

• ​To sort out RTP problems, I would recommend:

  1) on all endpoints use codec of allow=!all,ulaw — this is or should be supported by all endpoints and eliminates any issues of mismatch, translation, etc., and can be adjusted later once everything is working

  2) add an Echo() application to your dialplan so you can call it and check
  ​RTP to and from Asterisk

  3) start with direct_media=no to run all the RTP through Asterisk first

  4) packet capture at/on the asterisk server, as well as at endpoints if need be, to identfy if and where RTP streams are being sent and received.

  The goal being to get two way audio calls up through Asterisk, and then change one thing at a time towards your desired configuration and retest.