Pjsip rewrite port in from/to headers on reply

Asterisk Asterisk SIP
1

Hello! I have problems configuring chan_pjsip. In response to requests from my softswitch, the asterisk rewrites the headers from and to the port, removes the port from there. Tell me how to make it so that the asterisk does not do this? This problem is not observed for the chan_sip driver.

2019/02/13 10:31:09.351381 85.12.217.3:5060 -> 10.227.228.4:5061
INVITE sip:73432875561@10.227.228.4:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 85.12.217.3:5060;rport;branch=z9hG4bK-8ae2b6262f5011e99bf22c44fd84c5e0;sig=35ee398c
Via: SIP/2.0/UDP 85.12.217.3:5060;rport;branch=z9hG4bK-8ae2b0362f5011e99bf22c44fd84c5e0;sig=e4f91997
Via: SIP/2.0/UDP 85.12.217.3:5062;rport=5062;branch=z9hG4bK-8ae293762f5011e99bf22c44fd84c5e0
From: <sip:73432786060@85.12.217.3:5062;user=phone>;tag=8ae26efa2f5011e99bf22c44fd84c5e0**
To: <sip:73432875561@10.227.228.4:5061;user=phone>**
Call-ID: 8ae26f222f5011e99bf22c44fd84c5e0@85.12.217.3
CSeq: 1 INVITE
Contact: <sip:73432786060@85.12.217.3:5062;user=phone>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE
Max-Forwards: 69
User-Agent: TS-v4.6.0-07a
Cisco-Guid: 2329864784-793776617-2616339524-4253337056
Content-Length:   312
Record-Route: <sip:AQEAEOAgaLvZEqxL3Z9L9avyD5MDAAQ6GnBx@85.12.217.3;lr>
Record-Route: <sip:AQEAEEiS5FS8Xjpm4lEkLYOwprwDAARDhHft@85.12.217.3;lr>

v=0
o=- 1550035858 1550035858 IN IP4 85.12.217.17
s=-
c=IN IP4 85.12.217.17
t=0 0
m=audio 27044 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

2019/02/13 10:31:09.352290 10.227.228.4:5061 -> 85.12.217.3:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 85.12.217.3:5060;rport=5060;received=85.12.217.3;branch=z9hG4bK-8ae2b6262f5011e99bf22c44fd84c5e0;sig=35ee398c
Via: SIP/2.0/UDP 85.12.217.3:5060;rport;branch=z9hG4bK-8ae2b0362f5011e99bf22c44fd84c5e0;sig=e4f91997
Via: SIP/2.0/UDP 85.12.217.3:5062;rport=5062;branch=z9hG4bK-8ae293762f5011e99bf22c44fd84c5e0
Record-Route: <sip:AQEAEOAgaLvZEqxL3Z9L9avyD5MDAAQ6GnBx@85.12.217.3;lr>
Record-Route: <sip:AQEAEEiS5FS8Xjpm4lEkLYOwprwDAARDhHft@85.12.217.3;lr>
Call-ID: 8ae26f222f5011e99bf22c44fd84c5e0@85.12.217.3
From: <sip:73432786060@85.12.217.3;user=phone>;tag=8ae26efa2f5011e99bf22c44fd84c5e0
To: <sip:73432875561@10.227.228.4;user=phone>;tag=z9hG4bK-8ae2b6262f5011e99bf22c44fd84c5e0
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="centrex",nonce="1550035869/1ef0271eb3e86940bce3fc40fc143ef5",opaque="6f75f93b26037d9b",algorithm=md5,qop="auth"
Server: centrex
Content-Length:  0
2

What’s the endpoint configuration you are using?

3

Full config:

[global]
type=global
user_agent=centrex
default_from_user=centrex
default_realm=centrex

[transport-udp]
type=transport
protocol=udp    ;udp,tcp,tls,ws,wss
bind=0.0.0.0:5061

[opensips-endpoint](!)
type=endpoint
send_pai=no
send_rpid=no
;send_rpid=yes
rpid_immediate=no
trust_id_outbound=no
trust_id_inbound=no
context=from_opensips
from_domain=sbc.profintel.ru
sdp_session=Centrex v.1.0
disallow=all
allow=alaw
allow=ulaw
t38_udptl_maxdatagram=20
t38_udptl=yes
;---nat
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

[opensips-auth](!)
type=auth
auth_type=userpass
realm=centrex

[opensips-aor](!)
type=aor
max_contacts=1


[opensips_sbc1](opensips-aor)
[opensips_sbc1](opensips-auth)
password=oemae8angoc4utoh2aeP
username=opensips_sbc1
[opensips_sbc1](opensips-endpoint)
auth=opensips_sbc1
aors=opensips_sbc1
accountcode = opensips_sbc1

[opensips_sbc2](opensips-aor)
[opensips_sbc2](opensips-auth)
password=oemae8angoc4utoh2aeP
username=opensips_sbc2
[opensips_sbc2](opensips-endpoint)
auth=opensips_sbc2
aors=opensips_sbc2
accountcode = opensips_sbc2

[opensips_sbc2]
type=identify
endpoint=opensips_sbc2
match=10.227.228.18

[opensips_sbc1]
type=identify
endpoint=opensips_sbc1
match=10.227.228.169
4

Try setting “rewrite_contact” to no.

5

This is the first thing I tried, the result is negative.

6

I’m not sure then, it may be something in the PJSIP core code but I don’t know where it would be… or what.

7

This is a bug or feature. Who can I ask for help from?

8

They have a mailing list where things can be discussed, you could also file an issue on our issue tracker[1] but you’d need to ensure you are running the latest version of a supported branch and also provide the logs and full configuration. There’s also no time frame on when it would get looked into.

[1] https://issues.asterisk.org/jira

9

Thanks, I create issue on your tracker and will wait for an answer.

10

Port is not allowed in To and From as per RFC 3261. I tried pasting the relevant bit here, but the formatting is not good, so just read the last two paragraphs of 19.1.1 and then reference the chart in 19.1.2.

1 Like
11

Nice! I’ve never run into such a thing or really looked so didn’t know of that part of it. I’m surprised the parser didn’t consider it invalid, then. Usually it does for such things but it may be forgiving there.