Dear all,
I have an on-reply route that needs to change the SDP version for the reply coming in. The use case is that I have a mobile originated call and there is some Ericsson switch that doesn't like it when the SDP version is updated (in this case by asterisk) although nothing has changed to the actual SDP (183 session progress and then OK.) Funny thing is that Asterisk will actually drop a call if it receives a re-INVITE with same version... That's why they invented ignoresdpversion, but now it's the other way around.... :)
Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio (10.41.0.21) -> Asterisk
The issue is that the asterisk sends a reply 200 OK, with an updated version because it already sent SDP for 183 session progress. This can be patched in asterisk, but in my scenario I can unfortunately not do that. Thus trying to fix this on Kamailio.
I am able to 'fix' this currently by performing a subst on the sdp owner variable:
onreply_route[WITHSDP] { if (has_body("application/sdp")) { if(ds_is_from_list()) { rtpproxy_answer("wrei"); *if(subst("/^o=someowner ([0-9]+) ([0-9]+) IN IP4 (.*)$/o=someowner \1 \1 IN IP4 \3/")) {* xlog("L_INFO", "Fixed Asterisk incorrect version number in SDP"); } # tried the answer here as well, but that corrupts it even more } else { rtpproxy_answer("wrie"); } } exit; }
However this corrupts the SDP:
v=0 o=tismi 652858233 652858233 IN IP4 10.41.0.21*18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 57644 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes
When I do not substitute the SDP looks perfectly fine and the external address shows as the IN IP4. But of course the version is incremented:
v=0 o=tismi 1606876535 *1606876536* IN IP4 *18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 55410 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=sdpmangled:yes
The ericsson is now accepting this (although it's corrupt, I know.... probably the ACME doing something funky with it), but it causes issues with another unknown piece of equipment that fails on parsing the session owner. I hope there is something wrong with my subst, but I'm afraid I can not do this from the on_reply route because SDP is only updated once it finishes?
I know it's dangerous to alter the session version like this, so I made sure the Asterisk will never send a re-INVITE. Now I need a way to not corrupt the o=
Kind regards,
Matthias van der Vlies
Hello,
use msg_apply_changes() from textops modules in between the changes done to sdp.
No time to check the code, so I am not sure at this moment if can be done inside the onreply_route[x], which is executed by tm, but should be possible in core reply_route - -there you can make a filter and detect it is sdp, do the subst, apply the changes, then let the rtpproxy in onreply_route[x] like now.
Cheers, Daniel
On 15/01/15 13:38, Matthias van der Vlies wrote:
Dear all,
I have an on-reply route that needs to change the SDP version for the reply coming in. The use case is that I have a mobile originated call and there is some Ericsson switch that doesn't like it when the SDP version is updated (in this case by asterisk) although nothing has changed to the actual SDP (183 session progress and then OK.) Funny thing is that Asterisk will actually drop a call if it receives a re-INVITE with same version... That's why they invented ignoresdpversion, but now it's the other way around.... :)
Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio (10.41.0.21) -> Asterisk
The issue is that the asterisk sends a reply 200 OK, with an updated version because it already sent SDP for 183 session progress. This can be patched in asterisk, but in my scenario I can unfortunately not do that. Thus trying to fix this on Kamailio.
I am able to 'fix' this currently by performing a subst on the sdp owner variable:
onreply_route[WITHSDP] { if (has_body("application/sdp")) { if(ds_is_from_list()) { rtpproxy_answer("wrei"); *if(subst("/^o=someowner ([0-9]+) ([0-9]+) IN IP4 (.*)$/o=someowner \1 \1 IN IP4 \3/")) {* xlog("L_INFO", "Fixed Asterisk incorrect version number in SDP"); } # tried the answer here as well, but that corrupts it even more } else { rtpproxy_answer("wrie"); } } exit; }
However this corrupts the SDP:
v=0 o=tismi 652858233 652858233 IN IP4 10.41.0.21*18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 57644 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes
When I do not substitute the SDP looks perfectly fine and the external address shows as the IN IP4. But of course the version is incremented:
v=0 o=tismi 1606876535 *1606876536* IN IP4 *18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 55410 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=sdpmangled:yes
The ericsson is now accepting this (although it's corrupt, I know.... probably the ACME doing something funky with it), but it causes issues with another unknown piece of equipment that fails on parsing the session owner. I hope there is something wrong with my subst, but I'm afraid I can not do this from the on_reply route because SDP is only updated once it finishes?
I know it's dangerous to alter the session version like this, so I made sure the Asterisk will never send a re-INVITE. Now I need a way to not corrupt the o=
Kind regards,
Matthias van der Vlies
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On Thursday 15 January 2015 13:38:00 Matthias van der Vlies wrote:
I have an on-reply route that needs to change the SDP version for the reply coming in. The use case is that I have a mobile originated call and there is some Ericsson switch that doesn't like it when the SDP version is updated (in this case by asterisk) although nothing has changed to the actual SDP (183 session progress and then OK.) Funny thing is that Asterisk will actually drop a call if it receives a re-INVITE with same version... That's why they invented ignoresdpversion, but now it's the other way around.... :)
Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio (10.41.0.21) -> Asterisk
Sounds familiar (the Ericsson is the troublemaker), my solution was to upgrade to asterisk 11.x. But since patching is not possible, I assume upgrading is neither. A workaround was to drop the 183 message in kamailio to Ericsson:
onreply_route[MANAGE_REPLY] { xlog("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE);
if(status=="183") { drop; }
...
That's also a solution, but I prefer not dropping the 183 :)
Daniel's (the other one ;-)) suggestion was the solution in the end:
if(subst("/^o=somestuff ([0-9]+) ([0-9]+) IN IP4 (.*)$/o=somestuff \1 \1 IN IP4 \3/")) { xlog("L_INFO", "Fixed Asterisk incorrect version number in SDP"); msg_apply_changes(); } rtpproxy_answer("wrei");
Now it's looking OK:
v=0 o=tismi 895007076 895007076 IN IP4 185.X.X.X s=XXXXX c=IN IP4 185.X.X.X t=0 0 m=audio 39250 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=sdpmangled:yes
Matthias
On 01/15/2015 03:29 PM, Daniel Tryba wrote:
On Thursday 15 January 2015 13:38:00 Matthias van der Vlies wrote:
I have an on-reply route that needs to change the SDP version for the reply coming in. The use case is that I have a mobile originated call and there is some Ericsson switch that doesn't like it when the SDP version is updated (in this case by asterisk) although nothing has changed to the actual SDP (183 session progress and then OK.) Funny thing is that Asterisk will actually drop a call if it receives a re-INVITE with same version... That's why they invented ignoresdpversion, but now it's the other way around.... :)
Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio (10.41.0.21) -> Asterisk
Sounds familiar (the Ericsson is the troublemaker), my solution was to upgrade to asterisk 11.x. But since patching is not possible, I assume upgrading is neither. A workaround was to drop the 183 message in kamailio to Ericsson:
onreply_route[MANAGE_REPLY] { xlog("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE);
if(status=="183") { drop; }...
-
Daniel Tryba -
Daniel-Constantin Mierla -
Matthias van der Vlies