RTP problem with setup asterisk <-> openser <->

high.*!

i'm trying to get the interworking between OCS 2007 and asterisk (1.2.23) done and i chose openser (1.2.x) as interface between the two systems.
for basic setup i used the following guide http://confluence.terena.org:8080/display/IPTelCB/3.2.7.+Tying+MS+OCS+with+Asterisk+through+OpenSER

adoptions i need to do was to correct the 'CONTACT' part in the SIP headers in the request and the reply route to use the correct transport and maddr.
after doing this modifications in the SIP header SIP communication's working fine.

but as already stated in the above mentioned guide there is a problem with the RTP stream. the RTP stream should go directly from OCS mediaton server to asterisk.

in case of setting up a call in the office communicator it seems that the RTP stream is tried to be established between OCSmed and openser, which openser replies with an ICMP message "Destination not reachable".

looking at the (wireshark trace) it seems, that immediately after sending the INVITE request from OCSmed to openser a RTP request with wrong dst_ip (openser) and invalid dst_port (0) is triggered.

after receiving the SIP/SDP reply "200 OK" a RTCP (receiver report) message is sent from OCSmed(172.23.26.121) to asterisk(172.29.223.210), which seems correct, but there is no answer from asterisk towards OCSmed.
RTP traffic between asterisk and OCSmed isn't done at all?

does anyone has a hint solving my problem?

i tried now to involve openSER into the RTP traffic by using the nathelper module together with rtpproxy.
RTP traffic is now route via openSER but it didn't change something on the problem.

it seems to me that asterisk has a problem to establish the RTP communication. in the trace i see that openser (resp. the mediation server) is sending RTCP messages to asterisk but asterisk itself did no response.

does anyone has faced the same issue? any ideas how i can debug this RTP problem (on asterisk)?

thx & cheers
-hugo

finally i solved the RTP problem which was caused by the OCS mediation server appending "charset=utf-8" in the SIP header  line "content-type", which confuses asterisk.

so i simply removed this additional info and here we go... RTP's working!

my above mentioned rewriting of the "contact" header line, i simply solved by completely removing this header line. this is working fine too...

see here my updates to openser.cfg:

Code:

route { ...     if (src_ip == <IP_of_OCS_mediation_server>) { #       remove misleading CONTACT header line         remove_hf("Contact"); #       remove UTF-8 information, as * is not able to process it properly         subst("/^(CONTENT-TYPE:.*);[ ]*charset=utf-8(.*)/\1\2/"); #       relay request to *         if (!t_relay("udp:<IP_of_asterisk_server>:5060")) {             sl_reply_error();         }     }     else { #       relay request to OCS         if (!t_relay("tcp:<IP_of_OCS_mediation_server>:5060")  ) {             sl_reply_error();         }     } ... } on_reply_route { ...     if (src_ip == <IP_of_OCS_mediation_server>) { #       remove misleading CONTACT header line         remove_hf("Contact"); #       remove UTF-8 information, as * is not able to process it properly         subst("/^(CONTENT-TYPE:.*);[ ]*charset=utf-8(.*)/\1\2/");     } ... }

cheers

-hugo

nice one, and thanks for posting your findings once you'd solved your problem.. much appreciated!

Hugo,

I have been following your guide at http://confluence.terena.org:8080/displ ... gh+OpenSER , but I am stuck.

Can you please post your up to date openser.cfg, right now all my calls from OCS to Asterisk gets stuck on Openser with error too many hops:

tcp_send: buf= SIP/2.0 483 Too Many Hops^M FROM: <sip:+1234@ocsmediation.adrd.domain.com;user=phone>;epid=BC06E2A83B;tag=f63701b3^M TO: <sip:+14169550892@10.4.7.101;user=phone>;tag=8cd747c0cd8f42ddf4a044d77f697388.2c07^M CSEQ: 1 INVITE^M CALL-ID: 6b7dba04-87fc-4494-9e53-280767f17f6a^M VIA: SIP/2.0/TCP 10.4.21.3:2350;branch=z9hG4bK46e6797^M Server: OpenSER (1.2.1-notls (i386/linux))^M Content-Length: 0^M ^M

I tried to piece together openser.cfg on your article and your post here, but on your article there is no "on_reply_route { " section.

I had my setup working with OCS<>Sipx<>Asterisk before and now trying to make it work with OpenSer, but it is failing.

Your help is appreciated.